draft-ietf-rtcweb-jsep-03.txt   draft-ietf-rtcweb-jsep-04.txt 
Network Working Group J. Uberti Network Working Group J. Uberti
Internet-Draft Google Internet-Draft Google
Intended status: Standards Track C. Jennings Intended status: Standards Track C. Jennings
Expires: August 29, 2013 Cisco Expires: March 22, 2014 Cisco
February 25, 2013 September 18, 2013
Javascript Session Establishment Protocol Javascript Session Establishment Protocol
draft-ietf-rtcweb-jsep-03 draft-ietf-rtcweb-jsep-04
Abstract Abstract
This document describes the mechanisms for allowing a Javascript This document describes the mechanisms for allowing a Javascript
application to fully control the signaling plane of a multimedia application to control the signaling plane of a multimedia session
session via the interface specified in the W3C RTCPeerConnection API, via the interface specified in the W3C RTCPeerConnection API, and
and discusses how this relates to existing signaling protocols. discusses how this relates to existing signaling protocols.
Status of this Memo Status of This Memo
This Internet-Draft is submitted in full conformance with the This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79. provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute Task Force (IETF). Note that other groups may also distribute
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Internet-Drafts are draft documents valid for a maximum of six months Internet-Drafts are draft documents valid for a maximum of six months
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time. It is inappropriate to use Internet-Drafts as reference time. It is inappropriate to use Internet-Drafts as reference
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This Internet-Draft will expire on August 29, 2013. This Internet-Draft will expire on March 22, 2014.
Copyright Notice Copyright Notice
Copyright (c) 2013 IETF Trust and the persons identified as the Copyright (c) 2013 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
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Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
1.1. General Design of JSEP . . . . . . . . . . . . . . . . . . 4 1.1. General Design of JSEP . . . . . . . . . . . . . . . . . 3
1.2. Other Approaches Considered . . . . . . . . . . . . . . . 6 1.2. Other Approaches Considered . . . . . . . . . . . . . . . 5
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 6 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 6
3. Semantics and Syntax . . . . . . . . . . . . . . . . . . . . . 7 3. Semantics and Syntax . . . . . . . . . . . . . . . . . . . . 6
3.1. Signaling Model . . . . . . . . . . . . . . . . . . . . . 7 3.1. Signaling Model . . . . . . . . . . . . . . . . . . . . . 6
3.2. Session Descriptions and State Machine . . . . . . . . . . 7 3.2. Session Descriptions and State Machine . . . . . . . . . 7
3.3. Session Description Format . . . . . . . . . . . . . . . . 10 3.3. Session Description Format . . . . . . . . . . . . . . . 9
3.4. ICE . . . . . . . . . . . . . . . . . . . . . . . . . . . 10 3.4. ICE . . . . . . . . . . . . . . . . . . . . . . . . . . . 10
3.4.1. ICE Candidate Trickling . . . . . . . . . . . . . . . 10 3.4.1. ICE Candidate Trickling . . . . . . . . . . . . . . . 10
3.4.1.1. ICE Candidate Format . . . . . . . . . . . . . . . 11 3.4.1.1. ICE Candidate Format . . . . . . . . . . . . . . 10
3.5. Interactions With Forking . . . . . . . . . . . . . . . . 11 3.5. Interactions With Forking . . . . . . . . . . . . . . . . 11
3.5.1. Sequential Forking . . . . . . . . . . . . . . . . . . 12 3.5.1. Sequential Forking . . . . . . . . . . . . . . . . . 11
3.5.2. Parallel Forking . . . . . . . . . . . . . . . . . . . 12 3.5.2. Parallel Forking . . . . . . . . . . . . . . . . . . 12
3.6. Session Rehydration . . . . . . . . . . . . . . . . . . . 13 3.6. Session Rehydration . . . . . . . . . . . . . . . . . . . 13
4. Interface . . . . . . . . . . . . . . . . . . . . . . . . . . 14 4. Interface . . . . . . . . . . . . . . . . . . . . . . . . . . 13
4.1. SDP Requirements . . . . . . . . . . . . . . . . . . . . . 14 4.1. Methods . . . . . . . . . . . . . . . . . . . . . . . . . 14
4.2. Methods . . . . . . . . . . . . . . . . . . . . . . . . . 15 4.1.1. createOffer . . . . . . . . . . . . . . . . . . . . . 14
4.2.1. createOffer . . . . . . . . . . . . . . . . . . . . . 15 4.1.2. createAnswer . . . . . . . . . . . . . . . . . . . . 15
4.2.2. createAnswer . . . . . . . . . . . . . . . . . . . . . 16 4.1.3. SessionDescriptionType . . . . . . . . . . . . . . . 15
4.2.3. SessionDescriptionType . . . . . . . . . . . . . . . . 17 4.1.3.1. Use of Provisional Answers . . . . . . . . . . . 16
4.2.3.1. Use of Provisional Answers . . . . . . . . . . . . 18 4.1.3.2. Rollback . . . . . . . . . . . . . . . . . . . . 17
4.2.3.2. Rollback . . . . . . . . . . . . . . . . . . . . . 18 4.1.4. setLocalDescription . . . . . . . . . . . . . . . . . 17
4.2.4. setLocalDescription . . . . . . . . . . . . . . . . . 19 4.1.5. setRemoteDescription . . . . . . . . . . . . . . . . 18
4.2.5. setRemoteDescription . . . . . . . . . . . . . . . . . 19 4.1.6. localDescription . . . . . . . . . . . . . . . . . . 18
4.2.6. localDescription . . . . . . . . . . . . . . . . . . . 20 4.1.7. remoteDescription . . . . . . . . . . . . . . . . . . 18
4.2.7. remoteDescription . . . . . . . . . . . . . . . . . . 20 4.1.8. updateIce . . . . . . . . . . . . . . . . . . . . . . 19
4.2.8. updateIce . . . . . . . . . . . . . . . . . . . . . . 20 4.1.9. addIceCandidate . . . . . . . . . . . . . . . . . . . 19
4.2.9. addIceCandidate . . . . . . . . . . . . . . . . . . . 21 5. SDP Interaction Procedures . . . . . . . . . . . . . . . . . 19
5. SDP Interaction Procedures . . . . . . . . . . . . . . . . . . 21 5.1. SDP Requirements Overview . . . . . . . . . . . . . . . . 19
5.1. Constructing an Offer . . . . . . . . . . . . . . . . . . 21 5.2. Constructing an Offer . . . . . . . . . . . . . . . . . . 21
5.2. Generating an Answer . . . . . . . . . . . . . . . . . . . 21 5.2.1. Initial Offers . . . . . . . . . . . . . . . . . . . 21
5.3. Parsing an Offer . . . . . . . . . . . . . . . . . . . . . 21 5.2.2. Subsequent Offers . . . . . . . . . . . . . . . . . . 25
5.4. Parsing an Answer . . . . . . . . . . . . . . . . . . . . 21 5.2.3. Constraints Handling . . . . . . . . . . . . . . . . 26
5.5. Applying a Local Description . . . . . . . . . . . . . . . 21 5.2.3.1. OfferToReceiveAudio . . . . . . . . . . . . . . . 26
5.6. Applying a Remote Description . . . . . . . . . . . . . . 21 5.2.3.2. OfferToReceiveVideo . . . . . . . . . . . . . . . 27
6. Configurable SDP Parameters . . . . . . . . . . . . . . . . . 21 5.2.3.3. VoiceActivityDetection . . . . . . . . . . . . . 27
7. Security Considerations . . . . . . . . . . . . . . . . . . . 22 5.2.3.4. IceRestart . . . . . . . . . . . . . . . . . . . 27
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 23 5.3. Generating an Answer . . . . . . . . . . . . . . . . . . 27
9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 23 5.3.1. Initial Answers . . . . . . . . . . . . . . . . . . . 27
10. References . . . . . . . . . . . . . . . . . . . . . . . . . . 23 5.3.2. Subsequent Answers . . . . . . . . . . . . . . . . . 31
10.1. Normative References . . . . . . . . . . . . . . . . . . . 23 5.3.3. Constraints Handling . . . . . . . . . . . . . . . . 31
10.2. Informative References . . . . . . . . . . . . . . . . . . 24 5.4. Parsing an Offer . . . . . . . . . . . . . . . . . . . . 31
Appendix A. JSEP Implementation Examples . . . . . . . . . . . . 25 5.5. Parsing an Answer . . . . . . . . . . . . . . . . . . . . 31
A.1. Example API Flows . . . . . . . . . . . . . . . . . . . . 25 5.6. Applying a Local Description . . . . . . . . . . . . . . 31
A.1.1. Call using ROAP . . . . . . . . . . . . . . . . . . . 26 5.7. Applying a Remote Description . . . . . . . . . . . . . . 31
A.1.2. Call using XMPP . . . . . . . . . . . . . . . . . . . 26
A.1.3. Adding video to a call, using XMPP . . . . . . . . . . 28 6. Configurable SDP Parameters . . . . . . . . . . . . . . . . . 31
A.1.4. Simultaneous add of video streams, using XMPP . . . . 28 7. Security Considerations . . . . . . . . . . . . . . . . . . . 33
A.1.5. Call using SIP . . . . . . . . . . . . . . . . . . . . 29 8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 33
A.1.6. Handling early media (e.g. 1-800-GO FEDEX), using 9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 33
SIP . . . . . . . . . . . . . . . . . . . . . . . . . 30 10. References . . . . . . . . . . . . . . . . . . . . . . . . . 33
Appendix B. Change log . . . . . . . . . . . . . . . . . . . . . 31 10.1. Normative References . . . . . . . . . . . . . . . . . . 33
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 32 10.2. Informative References . . . . . . . . . . . . . . . . . 35
Appendix A. JSEP Implementation Examples . . . . . . . . . . . . 36
A.1. Example API Flows . . . . . . . . . . . . . . . . . . . . 36
A.1.1. Call using ROAP . . . . . . . . . . . . . . . . . . . 36
A.1.2. Call using XMPP . . . . . . . . . . . . . . . . . . . 37
A.1.3. Adding video to a call, using XMPP . . . . . . . . . 38
A.1.4. Simultaneous add of video streams, using XMPP . . . . 39
A.1.5. Call using SIP . . . . . . . . . . . . . . . . . . . 40
A.1.6. Handling early media (e.g. 1-800-GO FEDEX), using SIP 40
A.2. Example Session Descriptions . . . . . . . . . . . . . . 41
A.2.1. createOffer . . . . . . . . . . . . . . . . . . . . . 41
A.2.2. createAnswer . . . . . . . . . . . . . . . . . . . . 43
Appendix B. Change log . . . . . . . . . . . . . . . . . . . . . 44
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 45
1. Introduction 1. Introduction
This document describes how the W3C WEBRTC RTCPeerConnection This document describes how the W3C WEBRTC RTCPeerConnection
interface[W3C.WD-webrtc-20111027] is used to control the setup, interface[W3C.WD-webrtc-20111027] is used to control the setup,
management and teardown of a multimedia session. management and teardown of a multimedia session.
1.1. General Design of JSEP 1.1. General Design of JSEP
The thinking behind WebRTC call setup has been to fully specify and The thinking behind WebRTC call setup has been to fully specify and
skipping to change at page 6, line 42 skipping to change at page 6, line 12
generating offers and answers out of the browser. Instead of generating offers and answers out of the browser. Instead of
providing createOffer/createAnswer methods within the browser, this providing createOffer/createAnswer methods within the browser, this
approach would instead expose a getCapabilities API which would approach would instead expose a getCapabilities API which would
provide the application with the information it needed in order to provide the application with the information it needed in order to
generate its own session descriptions. This increases the amount of generate its own session descriptions. This increases the amount of
work that the application needs to do; it needs to know how to work that the application needs to do; it needs to know how to
generate session descriptions from capabilities, and especially how generate session descriptions from capabilities, and especially how
to generate the correct answer from an arbitrary offer and the to generate the correct answer from an arbitrary offer and the
supported capabilities. While this could certainly be addressed by supported capabilities. While this could certainly be addressed by
using a library like the one mentioned above, it basically forces the using a library like the one mentioned above, it basically forces the
use of said library even for a simple example. Providing use of said library even for a simple example. Providing createOffer
createOffer/createAnswer avoids this problem, but still allows /createAnswer avoids this problem, but still allows applications to
applications to generate their own offers/answers (to a large extent) generate their own offers/answers (to a large extent) if they choose,
if they choose, using the description generated by createOffer as an using the description generated by createOffer as an indication of
indication of the browser's capabilities. the browser's capabilities.
2. Terminology 2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in [RFC2119]. document are to be interpreted as described in [RFC2119].
3. Semantics and Syntax 3. Semantics and Syntax
3.1. Signaling Model 3.1. Signaling Model
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fashion described by [RFC3264] (offer/answer) in order for both sides fashion described by [RFC3264] (offer/answer) in order for both sides
of the session to know how to conduct the session. JSEP provides of the session to know how to conduct the session. JSEP provides
mechanisms to create offers and answers, as well as to apply them to mechanisms to create offers and answers, as well as to apply them to
a session. However, the browser is totally decoupled from the actual a session. However, the browser is totally decoupled from the actual
mechanism by which these offers and answers are communicated to the mechanism by which these offers and answers are communicated to the
remote side, including addressing, retransmission, forking, and glare remote side, including addressing, retransmission, forking, and glare
handling. These issues are left entirely up to the application; the handling. These issues are left entirely up to the application; the
application has complete control over which offers and answers get application has complete control over which offers and answers get
handed to the browser, and when. handed to the browser, and when.
+-----------+ +-----------+ +-----------+ +-----------+
| Web App |<--- App-Specific Signaling -->| Web App | | Web App |<--- App-Specific Signaling -->| Web App |
+-----------+ +-----------+ +-----------+ +-----------+
^ ^ ^ ^
| SDP | SDP | SDP | SDP
V V V V
+-----------+ +-----------+ +-----------+ +-----------+
| Browser |<----------- Media ------------>| Browser | | Browser |<----------- Media ------------>| Browser |
+-----------+ +-----------+ +-----------+ +-----------+
Figure 1: JSEP Signaling Model Figure 1: JSEP Signaling Model
3.2. Session Descriptions and State Machine 3.2. Session Descriptions and State Machine
In order to establish the media plane, the user agent needs specific In order to establish the media plane, the user agent needs specific
parameters to indicate what to transmit to the remote side, as well parameters to indicate what to transmit to the remote side, as well
as how to handle the media that is received. These parameters are as how to handle the media that is received. These parameters are
determined by the exchange of session descriptions in offers and determined by the exchange of session descriptions in offers and
answers, and there are certain details to this process that must be answers, and there are certain details to this process that must be
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v | v | v | v |
+---------------+ | +---------------+ | +---------------+ | +---------------+ |
| |----/ | |----/ | |----/ | |----/
| | setLocal(PRANSWER) | | | | setLocal(PRANSWER) | |
| Remote-Offer |------------------- >| Local-Pranswer| | Remote-Offer |------------------- >| Local-Pranswer|
| | | | | | | |
| | | | | | | |
+---------------+ +---------------+ +---------------+ +---------------+
^ | | ^ | |
| | setLocal(ANSWER) | | | setLocal(ANSWER) |
setRemote(OFFER) | | | setRemote(OFFER) | |
| V setLocal(ANSWER) | | V setLocal(ANSWER) |
+---------------+ | +---------------+ |
| | | | | |
| | | | | |
| Stable |<---------------------------+ | Stable |<---------------------------+
| | | | | |
| | | | | |
+---------------+ setRemote(ANSWER) | +---------------+ setRemote(ANSWER) |
^ | | ^ | |
| | setLocal(OFFER) | | | setLocal(OFFER) |
setRemote(ANSWER)| | | setRemote(ANSWER) | |
| V | | V |
+---------------+ +---------------+ +---------------+ +---------------+
| | | | | | | |
| | setRemote(PRANSWER) | | | | setRemote(PRANSWER) | |
| Local-Offer |------------------- >|Remote-Pranswer| | Local-Offer |------------------- >|Remote-Pranswer|
| | | | | | | |
| |----\ | |----\ | |----\ | |----\
+---------------+ | +---------------+ | +---------------+ | +---------------+ |
^ | ^ | ^ | ^ |
| | | | | | | |
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However, to simplify Javascript processing, and provide for future However, to simplify Javascript processing, and provide for future
flexibility, the SDP syntax is encapsulated within a flexibility, the SDP syntax is encapsulated within a
SessionDescription object, which can be constructed from SDP, and be SessionDescription object, which can be constructed from SDP, and be
serialized out to SDP. If future specifications agree on a JSON serialized out to SDP. If future specifications agree on a JSON
format for session descriptions, we could easily enable this object format for session descriptions, we could easily enable this object
to generate and consume that JSON. to generate and consume that JSON.
Other methods may be added to SessionDescription in the future to Other methods may be added to SessionDescription in the future to
simplify handling of SessionDescriptions from Javascript. In the simplify handling of SessionDescriptions from Javascript. In the
meantime, it would be simple to write a Javascript library to perform meantime, Javascript libraries can be used to perform these
these manipulations. manipulations.
Note that most applications should be able to treat the
SessionDescriptions produced and consumed by these various API calls
as opaque blobs; that is, the application will not need to read or
change them. The W3C API will provide appropriate APIs to allow the
application to control various session parameters, which will provide
the necessary information to the browser about what sort of
SessionDescription to produce.
3.4. ICE 3.4. ICE
When a new ICE candidate is available, the ICE Agent will notify the When a new ICE candidate is available, the ICE Agent will notify the
application via a callback; these candidates will automatically be application via a callback; these candidates will automatically be
added to the local session description. When all candidates have added to the local session description. When all candidates have
been gathered, the callback will also be invoked to signal that the been gathered, the callback will also be invoked to signal that the
gathering process is complete. gathering process is complete.
3.4.1. ICE Candidate Trickling 3.4.1. ICE Candidate Trickling
Candidate trickling is a technique through which a caller may Candidate trickling is a technique through which a caller may
incrementally provide candidates to the callee after the initial incrementally provide candidates to the callee after the initial
offer has been dispatched; the semantics of "Trickle ICE" are defined offer has been dispatched; the semantics of "Trickle ICE" are defined
in [I-D.rescorla-mmusic-ice-trickle]. This process allows the callee in [I-D.ivov-mmusic-trickle-ice]. This process allows the callee to
to begin acting upon the call and setting up the ICE (and perhaps begin acting upon the call and setting up the ICE (and perhaps DTLS)
DTLS) connections immediately, without having to wait for the caller connections immediately, without having to wait for the caller to
to gather all possible candidates. This results in faster call gather all possible candidates. This results in faster call startup
startup in cases where gathering is not performed prior to initating in cases where gathering is not performed prior to initiating the
the call. call.
JSEP supports optional candidate trickling by providing APIs that JSEP supports optional candidate trickling by providing APIs that
provide control and feedback on the ICE candidate gathering process. provide control and feedback on the ICE candidate gathering process.
Applications that support candidate trickling can send the initial Applications that support candidate trickling can send the initial
offer immediately and send individual candidates when they get the offer immediately and send individual candidates when they get the
notified of a new candidate; applications that do not support this notified of a new candidate; applications that do not support this
feature can simply wait for the indication that gathering is feature can simply wait for the indication that gathering is
complete, and then create and send their offer, with all the complete, and then create and send their offer, with all the
candidates, at this time. candidates, at this time.
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using the new remote candidates for connectivity checks. using the new remote candidates for connectivity checks.
3.4.1.1. ICE Candidate Format 3.4.1.1. ICE Candidate Format
As with session descriptions, the syntax of the IceCandidate object As with session descriptions, the syntax of the IceCandidate object
provides some abstraction, but can be easily converted to and from provides some abstraction, but can be easily converted to and from
the SDP candidate lines. the SDP candidate lines.
The candidate lines are the only SDP information that is contained The candidate lines are the only SDP information that is contained
within IceCandidate, as they represent the only information needed within IceCandidate, as they represent the only information needed
that is not present in the initial offer (i.e. for trickle that is not present in the initial offer (i.e. for trickle
candidates). This information is carried with the same syntax as the candidates). This information is carried with the same syntax as the
"candidate-attribute" field defined for ICE. For example: "candidate-attribute" field defined for ICE. For example:
candidate:1 1 UDP 1694498815 192.0.2.33 10000 typ host candidate:1 1 UDP 1694498815 192.0.2.33 10000 typ host
The IceCandidate object also contains fields to indicate which m= The IceCandidate object also contains fields to indicate which m=
line it should be associated with. The m line can be identified in line it should be associated with. The m line can be identified in
one of two ways; either by a m-line index, or a MID. The m-line one of two ways; either by a m-line index, or a MID. The m-line
index is a zero-based index, referring to the Nth m-line in the SDP. index is a zero-based index, referring to the Nth m-line in the SDP.
The MID uses the "media stream identification", as defined in The MID uses the "media stream identification", as defined in
[RFC3388] , to identify the m-line. WebRTC implementations creating [RFC5888] , to identify the m-line. WebRTC implementations creating
an ICE Candidate object MUST populate both of these fields. an ICE Candidate object MUST populate both of these fields.
Implementations receiving an ICE Candidate object SHOULD use the MID Implementations receiving an ICE Candidate object SHOULD use the MID
if they implement that functionality, or the m-line index, if not. if they implement that functionality, or the m-line index, if not.
3.5. Interactions With Forking 3.5. Interactions With Forking
Some call signaling systems allow various types of forking where an Some call signaling systems allow various types of forking where an
SDP Offer may be provided to more than one device. For example, SIP SDP Offer may be provided to more than one device. For example, SIP
[RFC3261] defines both a "Parallel Search" and "Sequential Search". [RFC3261] defines both a "Parallel Search" and "Sequential Search".
Although these are primarily signaling level issues that are outside Although these are primarily signaling level issues that are outside
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to simultaneously exchange media with multiple peers, the flow is to simultaneously exchange media with multiple peers, the flow is
slightly more complex, but the Javascript application can follow the slightly more complex, but the Javascript application can follow the
strategy that [RFC3960] describes using UPDATE. (It is worth noting strategy that [RFC3960] describes using UPDATE. (It is worth noting
that use cases where this is the desired behavior are very unusual.) that use cases where this is the desired behavior are very unusual.)
The UPDATE approach allows the signaling to set up a separate media The UPDATE approach allows the signaling to set up a separate media
flow for each peer that it wishes to exchange media with. In JSEP, flow for each peer that it wishes to exchange media with. In JSEP,
this offer used in the UPDATE would be formed by simply creating a this offer used in the UPDATE would be formed by simply creating a
new PeerConnection and making sure that the same local media streams new PeerConnection and making sure that the same local media streams
have been added into this new PeerConnection. Then the new have been added into this new PeerConnection. Then the new
PeerConnection object would produce a SDP offer that could be used by PeerConnection object would produce a SDP offer that could be used by
the signaling to perform the UPDATE strategy discussed in [RFC3960] . the signaling to perform the UPDATE strategy discussed in [RFC3960].
As a result of sharing the media streams, the application will end up As a result of sharing the media streams, the application will end up
with N parallel PeerConnection sessions, each with a local and remote with N parallel PeerConnection sessions, each with a local and remote
description and their own local and remote addresses. The media flow description and their own local and remote addresses. The media flow
from these sessions can be managed by specifying SDP direction from these sessions can be managed by specifying SDP direction
attributes in the descriptions, or the application can choose to play attributes in the descriptions, or the application can choose to play
out the media from all sessions mixed together. Of course, if the out the media from all sessions mixed together. Of course, if the
application wants to only keep a single session, it can simply application wants to only keep a single session, it can simply
terminate the sessions that it no longer needs. terminate the sessions that it no longer needs.
skipping to change at page 13, line 43 skipping to change at page 13, line 27
than normal call signaling messages. than normal call signaling messages.
With rehydration, the current signaling state is persisted somewhere With rehydration, the current signaling state is persisted somewhere
outside of the page, perhaps on the application server, or in browser outside of the page, perhaps on the application server, or in browser
local storage. The page is then reloaded, the saved signaling state local storage. The page is then reloaded, the saved signaling state
is retrieved, and a new PeerConnection object is created for the is retrieved, and a new PeerConnection object is created for the
session. The previously obtained MediaStreams are re-acquired, and session. The previously obtained MediaStreams are re-acquired, and
are given the same IDs as the original session; this ensures the IDs are given the same IDs as the original session; this ensures the IDs
in use by the remote side continue to work. Next, a new offer is in use by the remote side continue to work. Next, a new offer is
generated by the new PeerConnection; this offer will have new ICE and generated by the new PeerConnection; this offer will have new ICE and
possibly new SDES credentials (since the old ICE and SRTP state has possibly new DTLS-SRTP certificate fingerprints (since the old ICE
been lost). Finally, this offer is used to re-initiate the session and SRTP state has been lost). Finally, this offer is used to re-
with the existing remote endpoint, who simply sees the new offer as initiate the session with the existing remote endpoint, who simply
an in-call renegotiation, and replies with an answer that can be sees the new offer as an in-call renegotiation, and replies with an
supplied to setRemoteDescription. ICE processing proceeds as usual, answer that can be supplied to setRemoteDescription. ICE processing
and as soon as connectivity is established, the session will be back proceeds as usual, and as soon as connectivity is established, the
up and running again. session will be back up and running again.
[OPEN ISSUE: EKR proposed an alternative rehydration approach where [OPEN ISSUE: EKR proposed an alternative rehydration approach where
the actual internal PeerConnection object in the browser was kept the actual internal PeerConnection object in the browser was kept
alive for some time after the web page was killed and provided some alive for some time after the web page was killed and provided some
way for a new page to acquire the old PeerConnection object.] way for a new page to acquire the old PeerConnection object.]
4. Interface 4. Interface
This section details the basic operations that must be present to This section details the basic operations that must be present to
implement JSEP functionality. The actual API exposed in the W3C API implement JSEP functionality. The actual API exposed in the W3C API
may have somewhat different syntax, but should map easily to these may have somewhat different syntax, but should map easily to these
concepts. concepts.
4.1. SDP Requirements 4.1. Methods
Note: The text in this section may not represent working group
consensus and is put here so that the working group can discuss it
and find out how to change it such that it does have consensus.
When generating SDP blobs, either for offers or answers, the
generated SDP needs to conform to the following specifications.
Similarly, in order to properly process received SDP blobs,
implementations need to implement the functionality described in the
following specifications. This list is derived from
[I-D.ietf-rtcweb-rtp-usage].
R-1 [RFC4566] is the base SDP specification and MUST be
implemented.
R-2 The [RFC5888] grouping framework MUST be implemented for
signaling grouping information, and MUST be used to identify m=
lines via the a=mid attribute.
R-3 [RFC5124] MUST be supported for signaling RTP/SAVPF RTP
profile.
R-4 [RFC4585] MUST be implemented to signal RTCP based feedback.
R-5 [RFC5245] MUST be implemented for signaling the ICE candidate
lines corresponding to each media stream.
R-6 [RFC5761] MUST be implemented to signal multiplexing of RTP and
RTCP.
R-7 The SDP atributes of "sendonly", "recvonly", "inactive", and
"sendrecv" from [RFC4566] MUST be implemented to signal
information about media direction.
R-8 [RFC5576] MUST be implemented to signal RTP SSRC values. [OPEN
ISSUE; depends on BUNDLE and how we choose to represent
multiple media sources]
R-9 [RFC5763] MUST be implemented to signal DTLS certificate
fingerprints.
R-10 [RFC5506] MAY be implemented to signal Reduced-Size RTCP
messages.
R-11 [RFC3556] with bandwidth modifiers MAY be supported for
specifying RTCP bandwidth as a fraction of the media bandwidth,
RTCP fraction allocated to the senders and setting maximum
media bit-rate boundaries.
R-12 [RFC4568] MAY be implemented to signal SDES SRTP keying
information.
R-13 A TBD-draft MUST be supported, in order to signal associations
between RTP objects and W3C MediaStreams and MediaStreamTracks
in a standard way. Though there is not yet WG consensus in
this area, this TBD-draft is very likely to be
[I-D.alvestrand-mmusic-msid].
R-14 A TBD-draft MUST be supported to signal the use or multiplexing
RTP somethings on a single UDP port, in order to avoid
excessive use of port number resources. Though there is not
yet WG consensus in this area, this TBD-draft is very likely to
be [I-D.holmberg-mmusic-sdp-bundle-negotiation].
As required by [RFC4566] Section 5.13 JSEP implementations MUST
ignore unknown attributes (a=) lines.
Example SDP for RTCWeb call flows can be found in
[I-D.nandakumar-rtcweb-sdp]. [TODO: since we are starting to
specify how to handle SDP in this document, should these call flows
be merged into this document, or this link moved to the examples
section?]
4.2. Methods
4.2.1. createOffer 4.1.1. createOffer
The createOffer method generates a blob of SDP that contains a The createOffer method generates a blob of SDP that contains a
[RFC3264] offer with the supported configurations for the session, [RFC3264] offer with the supported configurations for the session,
including descriptions of the local MediaStreams attached to this including descriptions of the local MediaStreams attached to this
PeerConnection, the codec/RTP/RTCP options supported by this PeerConnection, the codec/RTP/RTCP options supported by this
implementation, and any candidates that have been gathered by the ICE implementation, and any candidates that have been gathered by the ICE
Agent. A constraints parameters may be supplied to provide Agent. A constraints parameters may be supplied to provide
additional control over the generated offer. This constraints additional control over the generated offer. This constraints
parameter should allow for the following manipulations to be parameter should allow for the following manipulations to be
performed: performed:
skipping to change at page 16, line 12 skipping to change at page 14, line 33
o To trigger an ICE restart, for the purpose of reestablishing o To trigger an ICE restart, for the purpose of reestablishing
connectivity. connectivity.
o For re-offer cases, to request an offer that contains the full set o For re-offer cases, to request an offer that contains the full set
of supported capabilities, as opposed to just the currently of supported capabilities, as opposed to just the currently
negotiated parameters. negotiated parameters.
In the initial offer, the generated SDP will contain all desired In the initial offer, the generated SDP will contain all desired
functionality for the session (certain parts that are supported but functionality for the session (certain parts that are supported but
not desired by default may be omitted); for each SDP line, the not desired by default may be omitted); for each SDP line, the
generation of the SDP must follow the process defined for generating generation of the SDP will follow the process defined for generating
an initial offer from the document (listed in Section 4.1) that an initial offer from the document that specifies the given SDP line.
specifies the given SDP line. The exact handling of initial offer generation is detailed in
Section 5.2.1. below.
In the event createOffer is called after the session is established, In the event createOffer is called after the session is established,
createOffer will generate an offer to modify the current session createOffer will generate an offer to modify the current session
based on any changes that have been made to the session, e.g. adding based on any changes that have been made to the session, e.g. adding
or removing MediaStreams, or requesting an ICE restart. For each or removing MediaStreams, or requesting an ICE restart. For each
existing stream, the generation of each SDP line must follow the existing stream, the generation of each SDP line must follow the
process defined for generating an updated offer from the document process defined for generating an updated offer from the document
that specfies the given SDP line. For each new stream, the that specifies the given SDP line. For each new stream, the
generation of the SDP must follow the process of generating an generation of the SDP must follow the process of generating an
initial offer, as mentioned above. If no changes have been made, or initial offer, as mentioned above. If no changes have been made, or
for SDP lines that are unaffected by the requested changes, the offer for SDP lines that are unaffected by the requested changes, the offer
will only contain the parameters negotiated by the last offer-answer will only contain the parameters negotiated by the last offer-answer
exchange. exchange. The exact handling of subsequent offer generation is
detailed in Section 5.2.2. below.
Session descriptions generated by createOffer must be immediately Session descriptions generated by createOffer must be immediately
usable by setLocalDescription; if a system has limited resources usable by setLocalDescription; if a system has limited resources
(e.g. a finite number of decoders), createOffer should return an (e.g. a finite number of decoders), createOffer should return an
offer that reflects the current state of the system, so that offer that reflects the current state of the system, so that
setLocalDescription will succeed when it attempts to acquire those setLocalDescription will succeed when it attempts to acquire those
resources. Because this method may need to inspect the system state resources. Because this method may need to inspect the system state
to determine the currently available resources, it may be implemented to determine the currently available resources, it may be implemented
as an async operation. as an async operation.
Calling this method may do things such as generate new ICE Calling this method may do things such as generate new ICE
credentials, but does not result in candidate gathering, or cause credentials, but does not result in candidate gathering, or cause
media to start or stop flowing. media to start or stop flowing.
4.2.2. createAnswer 4.1.2. createAnswer
The createAnswer method generates a blob of SDP that contains a The createAnswer method generates a blob of SDP that contains a
[RFC3264] SDP answer with the supported configuration for the session [RFC3264] SDP answer with the supported configuration for the session
that is compatible with the parameters supplied in the offer. Like that is compatible with the parameters supplied in the offer. Like
createOffer, the returned blob contains descriptions of the local createOffer, the returned blob contains descriptions of the local
MediaStreams attached to this PeerConnection, the codec/RTP/RTCP MediaStreams attached to this PeerConnection, the codec/RTP/RTCP
options negotiated for this session, and any candidates that have options negotiated for this session, and any candidates that have
been gathered by the ICE Agent. A constraints parameter may be been gathered by the ICE Agent. A constraints parameter may be
supplied to provide additional control over the generated answer. supplied to provide additional control over the generated answer.
As an answer, the generated SDP will contain a specific configuration As an answer, the generated SDP will contain a specific configuration
that specifies how the media plane should be established; for each that specifies how the media plane should be established; for each
SDP line, the generation of the SDP must follow the process defined SDP line, the generation of the SDP must follow the process defined
for generating an answer from the document that specifies the given for generating an answer from the document that specifies the given
SDP line. SDP line. The exact handling of answer generation is detailed in
Section 5.3. below.
Session descriptions generated by createAnswer must be immediately Session descriptions generated by createAnswer must be immediately
usable by setLocalDescription; like createOffer, the returned usable by setLocalDescription; like createOffer, the returned
description should reflect the current state of the system. Because description should reflect the current state of the system. Because
this method may need to inspect the system state to determine the this method may need to inspect the system state to determine the
currently available resources, it may need to be implemented as an currently available resources, it may need to be implemented as an
async operation. async operation.
Calling this method may do things such as generate new ICE Calling this method may do things such as generate new ICE
credentials, but does not trigger candidate gathering or change media credentials, but does not trigger candidate gathering or change media
state. state.
4.2.3. SessionDescriptionType 4.1.3. SessionDescriptionType
Session description objects (RTCSessionDescription) may be of type Session description objects (RTCSessionDescription) may be of type
"offer", "pranswer", and "answer". These types provide information "offer", "pranswer", and "answer". These types provide information
as to how the description parameter should be parsed, and how the as to how the description parameter should be parsed, and how the
media state should be changed. media state should be changed.
"offer" indicates that a description should be parsed as an offer; "offer" indicates that a description should be parsed as an offer;
said description may include many possible media configurations. A said description may include many possible media configurations. A
description used as an "offer" may be applied anytime the description used as an "offer" may be applied anytime the
PeerConnection is in a stable state, or as an update to a previously PeerConnection is in a stable state, or as an update to a previously
skipping to change at page 18, line 11 skipping to change at page 16, line 36
were allocated as a result of the offer. As such, the application were allocated as a result of the offer. As such, the application
can use some discretion on whether an answer should be applied as can use some discretion on whether an answer should be applied as
provisional or final, and can change the type of the session provisional or final, and can change the type of the session
description as needed. For example, in a serial forking scenario, an description as needed. For example, in a serial forking scenario, an
application may receive multiple "final" answers, one from each application may receive multiple "final" answers, one from each
remote endpoint. The application could choose to accept the initial remote endpoint. The application could choose to accept the initial
answers as provisional answers, and only apply an answer as final answers as provisional answers, and only apply an answer as final
when it receives one that meets its criteria (e.g. a live user when it receives one that meets its criteria (e.g. a live user
instead of voicemail). instead of voicemail).
4.2.3.1. Use of Provisional Answers 4.1.3.1. Use of Provisional Answers
Most web applications will not need to create answers using the Most web applications will not need to create answers using the
"pranswer" type. The preferred handling for a web application would "pranswer" type. The preferred handling for a web application would
be to create and send an "inactive" answer more or less immediately be to create and send an "inactive" answer more or less immediately
after receiving the offer, instead of waiting for a human user to after receiving the offer, instead of waiting for a human user to
physically answer the call. Later, when the human input is received, physically answer the call. Later, when the human input is received,
the application can create a new "sendrecv" offer to update the the application can create a new "sendrecv" offer to update the
previous offer/answer pair and start the media flow. This approach previous offer/answer pair and start the media flow. This approach
is preferred because it minimizes the amount of time that the offer- is preferred because it minimizes the amount of time that the offer-
answer exchange is left open, in addition to avoiding media clipping answer exchange is left open, in addition to avoiding media clipping
by ensuring the transport is ready to go by the time the call is by ensuring the transport is ready to go by the time the call is
phyiscally answered. However, some applications may not be able to physically answered. However, some applications may not be able to
do this, particularly ones that are attempting to gateway to other do this, particularly ones that are attempting to gateway to other
signaling protocols. In these cases, "pranswer" can still allow the signaling protocols. In these cases, "pranswer" can still allow the
application to warm up the transport. application to warm up the transport.
Consider a typical web application that will set up a data channel, Consider a typical web application that will set up a data channel,
an audio channel, and a video channel. When an endpoint receives an an audio channel, and a video channel. When an endpoint receives an
offer with these channels, it could send an answer accepting the data offer with these channels, it could send an answer accepting the data
channel for two-way data, and accepting the audio and video tracks as channel for two-way data, and accepting the audio and video tracks as
inactive or receive-only. It could then ask the user to accept the inactive or receive-only. It could then ask the user to accept the
call, acquire the local media streams, and send a new offer to the call, acquire the local media streams, and send a new offer to the
remote side moving the audio and video to be two-way media. By the remote side moving the audio and video to be two-way media. By the
time the human has accepted the call and sent the new offer, it is time the human has accepted the call and sent the new offer, it is
likely that the ICE and DTLS handshaking for all the channels will likely that the ICE and DTLS handshaking for all the channels will
already be set up. already be set up.
4.2.3.2. Rollback 4.1.3.2. Rollback
In certain situations it may be desirable to "undo" a change made to In certain situations it may be desirable to "undo" a change made to
setLocalDescription or setRemoteDescription. Consider a case where a setLocalDescription or setRemoteDescription. Consider a case where a
call is ongoing, and one side wants to change some of the session call is ongoing, and one side wants to change some of the session
parameters; that side generates an updated offer and then calls parameters; that side generates an updated offer and then calls
setLocalDescription. However, the remote side, either before or setLocalDescription. However, the remote side, either before or
after setRemoteDescription, decides it does not want to accept the after setRemoteDescription, decides it does not want to accept the
new parameters, and sends a reject message back to the offerer. Now, new parameters, and sends a reject message back to the offerer. Now,
the offerer, and possibly the answerer as well, need to return to a the offerer, and possibly the answerer as well, need to return to a
stable state and the previous local/remote description. To support stable state and the previous local/remote description. To support
skipping to change at page 19, line 17 skipping to change at page 17, line 41
candidates that were allocated by the new local description are candidates that were allocated by the new local description are
discarded; any media that is received will be processed according to discarded; any media that is received will be processed according to
the previous session description. the previous session description.
A rollback is performed by supplying a session description of type A rollback is performed by supplying a session description of type
"rollback" to either setLocalDescription or setRemoteDescription, "rollback" to either setLocalDescription or setRemoteDescription,
depending on which needs to be rolled back (i.e. if the new offer was depending on which needs to be rolled back (i.e. if the new offer was
supplied to setLocalDescription, the rollback should be done on supplied to setLocalDescription, the rollback should be done on
setLocalDescription as well.) setLocalDescription as well.)
4.2.4. setLocalDescription 4.1.4. setLocalDescription
The setLocalDescription method instructs the PeerConnection to apply The setLocalDescription method instructs the PeerConnection to apply
the supplied SDP blob as its local configuration. The type field the supplied SDP blob as its local configuration. The type field
indicates whether the blob should be processed as an offer, indicates whether the blob should be processed as an offer,
provisional answer, or final answer; offers and answers are checked provisional answer, or final answer; offers and answers are checked
differently, using the various rules that exist for each SDP line. differently, using the various rules that exist for each SDP line.
This API changes the local media state; among other things, it sets This API changes the local media state; among other things, it sets
up local resources for receiving and decoding media. In order to up local resources for receiving and decoding media. In order to
successfully handle scenarios where the application wants to offer to successfully handle scenarios where the application wants to offer to
skipping to change at page 19, line 46 skipping to change at page 18, line 21
local description is supplied, and the number of transports currently local description is supplied, and the number of transports currently
in use does not match the number of transports needed by the local in use does not match the number of transports needed by the local
description, the PeerConnection will create transports as needed and description, the PeerConnection will create transports as needed and
begin gathering candidates for them. begin gathering candidates for them.
If setRemoteDescription was previous called with an offer, and If setRemoteDescription was previous called with an offer, and
setLocalDescription is called with an answer (provisional or final), setLocalDescription is called with an answer (provisional or final),
and the media directions are compatible, and media are available to and the media directions are compatible, and media are available to
send, this will result in the starting of media transmission. send, this will result in the starting of media transmission.
4.2.5. setRemoteDescription 4.1.5. setRemoteDescription
The setRemoteDescription method instructs the PeerConnection to apply The setRemoteDescription method instructs the PeerConnection to apply
the supplied SDP blob as the desired remote configuration. As in the supplied SDP blob as the desired remote configuration. As in
setLocalDescription, the type field of the indicates how the blob setLocalDescription, the type field of the indicates how the blob
should be processed. should be processed.
This API changes the local media state; among other things, it sets This API changes the local media state; among other things, it sets
up local resources for sending and encoding media. up local resources for sending and encoding media.
If setRemoteDescription was previously called with an offer, and If setRemoteDescription was previously called with an offer, and
setLocalDescription is called with an answer (provisional or final), setLocalDescription is called with an answer (provisional or final),
and the media directions are compatible, and media are available to and the media directions are compatible, and media are available to
send, this will result in the starting of media transmission. send, this will result in the starting of media transmission.
4.2.6. localDescription 4.1.6. localDescription
The localDescription method returns a copy of the current local The localDescription method returns a copy of the current local
configuration, i.e. what was most recently passed to configuration, i.e. what was most recently passed to
setLocalDescription, plus any local candidates that have been setLocalDescription, plus any local candidates that have been
generated by the ICE Agent. generated by the ICE Agent.
TODO: Do we need to expose accessors for both the current and TODO: Do we need to expose accessors for both the current and
proposed local description? proposed local description?
A null object will be returned if the local description has not yet A null object will be returned if the local description has not yet
been established, or if the PeerConnection has been closed. been established, or if the PeerConnection has been closed.
4.2.7. remoteDescription 4.1.7. remoteDescription
The remoteDescription method returns a copy of the current remote The remoteDescription method returns a copy of the current remote
configuration, i.e. what was most recently passed to configuration, i.e. what was most recently passed to
setRemoteDescription, plus any remote candidates that have been setRemoteDescription, plus any remote candidates that have been
supplied via processIceMessage. supplied via processIceMessage.
TODO: Do we need to expose accessors for both the current and TODO: Do we need to expose accessors for both the current and
proposed remote description? proposed remote description?
A null object will be returned if the remote description has not yet A null object will be returned if the remote description has not yet
been established, or if the PeerConnection has been closed. been established, or if the PeerConnection has been closed.
4.2.8. updateIce 4.1.8. updateIce
The updateIce method allows the configuration of the ICE Agent to be The updateIce method allows the configuration of the ICE Agent to be
changed during the session, primarily for changing which types of changed during the session, primarily for changing which types of
local candidates are provided to the application and used for local candidates are provided to the application and used for
connectivity checks. A callee may initially configure the ICE Agent connectivity checks. A callee may initially configure the ICE Agent
to use only relay candidates, to avoid leaking location information, to use only relay candidates, to avoid leaking location information,
but update this configuration to use all candidates once the call is but update this configuration to use all candidates once the call is
accepted. accepted.
Regardless of the configuration, the gathering process collects all Regardless of the configuration, the gathering process collects all
available candidates, but excluded candidates will not be surfaced in available candidates, but excluded candidates will not be surfaced in
onicecandidate callback or used for connectivity checks. onicecandidate callback or used for connectivity checks.
This call may result in a change to the state of the ICE Agent, and This call may result in a change to the state of the ICE Agent, and
may result in a change to media state if it results in connectivity may result in a change to media state if it results in connectivity
being established. being established.
4.2.9. addIceCandidate 4.1.9. addIceCandidate
The addIceCandidate method provides a remote candidate to the ICE The addIceCandidate method provides a remote candidate to the ICE
Agent, which, if parsed successfully, will be added to the remote Agent, which, if parsed successfully, will be added to the remote
description according to the rules defined for Trickle ICE. description according to the rules defined for Trickle ICE.
Connectivity checks will be sent to the new candidate. Connectivity checks will be sent to the new candidate.
This call will result in a change to the state of the ICE Agent, and This call will result in a change to the state of the ICE Agent, and
may result in a change to media state if it results in connectivity may result in a change to media state if it results in connectivity
being established. being established.
5. SDP Interaction Procedures 5. SDP Interaction Procedures
This section describes the specific procedures to be followed when This section describes the specific procedures to be followed when
creating and parsing SDP objects. [Work In Progress] creating and parsing SDP objects.
5.1. Constructing an Offer 5.1. SDP Requirements Overview
The key specifications that govern creation and processing of offers
and answers are listed below. This list is derived from
[I-D.ietf-rtcweb-rtp-usage].
5.2. Generating an Answer R-1 [RFC4566] is the base SDP specification and MUST be
implemented.
5.3. Parsing an Offer R-2 The [RFC5888] grouping framework MUST be implemented for
signaling grouping information, and MUST be used to identify m=
lines via the a=mid attribute.
5.4. Parsing an Answer R-3 [RFC5124] MUST be supported for signaling RTP/SAVPF RTP
profile.
5.5. Applying a Local Description R-4 [RFC4585] MUST be implemented to signal RTCP based feedback.
5.6. Applying a Remote Description R-5 [RFC5245] MUST be implemented for signaling the ICE candidate
lines corresponding to each media stream.
R-6 [RFC5761] MUST be implemented to signal multiplexing of RTP and
RTCP.
R-7 The SDP atributes of "sendonly", "recvonly", "inactive", and
"sendrecv" from [RFC4566] MUST be implemented to signal
information about media direction.
R-8 [RFC5576] MUST be implemented to signal RTP SSRC values.
R-9 [RFC5763] MUST be implemented to signal DTLS certificate
fingerprints.
R-10 [RFC5506] MAY be implemented to signal Reduced-Size RTCP
messages.
R-11 [RFC3556] with bandwidth modifiers MAY be supported for
specifying RTCP bandwidth as a fraction of the media bandwidth,
RTCP fraction allocated to the senders and setting maximum media
bit-rate boundaries.
R-12 [RFC4568] MUST NOT be implemented to signal SDES SRTP keying
information.
R-13 A [I-D.ietf-mmusic-msid] MUST be supported, in order to signal
associations between RTP objects and W3C MediaStreams and
MediaStreamTracks in a standard way.
R-14 The bundle mechanism in
[I-D.ietf-mmusic-sdp-bundle-negotiation] MUST be supported to
signal the use or multiplexing RTP somethings on a single UDP
port, in order to avoid excessive use of port number resources.
As required by [RFC4566] Section 5.13 JSEP implementations MUST
ignore unknown attributes (a=) lines.
Example SDP for RTCWeb call flows can be found in
[I-D.nandakumar-rtcweb-sdp]. [TODO: since we are starting to specify
how to handle SDP in this document, should these call flows be merged
into this document, or this link moved to the examples section?]
5.2. Constructing an Offer
When createOffer is called, a new SDP description must be created
that includes the functionality specified in
[I-D.ietf-rtcweb-rtp-usage]. The exact details of this process are
explained below.
5.2.1. Initial Offers
When createOffer is called for the first time, the result is known as
the initial offer.
The first step in generating an initial offer is to generate session-
level attributes, as specified in [RFC4566], Section 5.
Specifically:
o The first SDP line MUST be "v=0", as specified in [RFC4566],
Section 5.1
o The second SDP line MUST be an "o=" line, as specified in
[RFC4566], Section 5.2. The value of the <username> field SHOULD
be "-". The value of the <sess-id> field SHOULD be a
cryptographically random number. To ensure uniqueness, this
number SHOULD be at least 64 bits long. The value of the <sess-
version> field SHOULD be zero. The value of the <nettype>
<addrtype> <unicast-address> tuple SHOULD be set to a non-
meaningful address, such as IN IP4 0.0.0.0, to prevent leaking the
local address in this field. As mentioned in [RFC4566], the
entire o= line needs to be unique, but selecting a random number
for <sess-id> is sufficient to accomplish this.
o The third SDP line MUST be a "s=" line, as specified in [RFC4566],
Section 5.3; a single space SHOULD be used as the session name,
e.g. "s= "
o Session Information ("i="), URI ("u="), Email Address ("e="),
Phone Number ("p="), Bandwidth ("b="), Repeat Times ("r="), and
Time Zones ("z=") lines are not useful in this context and SHOULD
NOT be included.
o Encryption Keys ("k=") lines do not provide sufficient security
and MUST NOT be included.
o A "t=" line MUST be added, as specified in [RFC4566], Section 5.9;
both <start-time> and <stop-time> SHOULD be set to zero, e.g. "t=0
0".
The next step is to generate m= sections for each MediaStreamTrack
that has been added to the PeerConnection via the addStream method.
Note that this method takes a MediaStream, which can contain multiple
MediaStreamTracks, and therefore multiple m= sections can be
generated even if addStream is only called once.
Each m= section should be generated as specified in [RFC4566],
Section 5.14. The <proto> field MUST be set to "RTP/SAVPF". If a m=
section is not being bundled into another m= section, it MUST
generate a unique set of ICE credentials and gather its own set of
candidates. Otherwise, it MUST use the same ICE credentials and
candidates that were used in the m= section that it is being bundled
into. For DTLS, all m= sections MUST use the same certificate [OPEN
ISSUE: how this is configured] and will therefore have the same
fingerprint values.
Each m= section MUST include the following:
o An "a=mid" line, as specified in [RFC5888], Section 4.
o An "a=msid" line, as specified in [I-D.ietf-mmusic-msid],
Section 2.
o [OPEN ISSUE: Use of App Token versus stream-correlator ]
o An "a=sendrecv" line, as specified in [RFC3264], Section 5.1.
o For each supported codec, "a=rtpmap" and "a=fmtp" lines, as
specified in [RFC4566], Section 6. For audio, the codecs
specified in [I-D.ietf-rtcweb-audio], Section 3, MUST be be
supported.
o For each primary codec where RTP retransmission should be used, a
corresponding "a=rtpmap" line indicating "rtx" with the clock rate
of the primary codec and an "a=fmtp" line that references the
payload type fo the primary codec, as specified in [RFC4588],
Section 8.1.
o For each supported FEC mechanism, a corresponding "a=rtpmap" line
indicating the desired FEC codec.
o "a=ice-ufrag" and "a=ice-passwd" lines, as specified in [RFC5245],
Section 15.4.
o An "a=ice-options" line, with the "trickle" option, as specified
in [I-D.ivov-mmusic-trickle-ice], Section 4.
o For each candidate that has been gathered during the most recent
gathering phase, an "a=candidate" line, as specified in [RFC5245],
Section 4.3., paragraph 3.
o For the current default candidate, a "c=" line, as specific in
[RFC5245], Section 4.3., paragraph 6. [OPEN ISSUE, pending
resolution in mmusic: If no candidates have yet been gathered yet,
the default candidate should be set to the null value defined in
[I-D.ivov-mmusic-trickle-ice], Section 5.1.]
o An "a=fingerprint" line, as specified in [RFC4572], Section 5.
Use of the SHA-256 algorithm for the fingerprint is REQUIRED; if
the browser also supports stronger hashes, additional
"a=fingerprint" lines with these hashes MAY also be added.
o An "a=setup" line, as specified in [RFC4145], Section 4, and
clarified for use in DTLS-SRTP scenarios in [RFC5763], Section 5.
The role value in the offer MUST be "actpass".
o An "a=rtcp-mux" line, as specified in [RFC5761], Section 5.1.1.
o An "a=rtcp-rsize" line, as specified in [RFC5506], Section 5.
o For each supported RTP header extension, an "a=extmap" line, as
specified in [RFC5285], Section 5. The list of header extensions
that SHOULD/MUST be supported is specified in
[I-D.ietf-rtcweb-rtp-usage], Section 5.2. Any header extensions
that require encryption MUST be specified as indicated in
[RFC6904], Section 4.
o For each supported RTCP feedback mechanism, an "a=rtcp-fb"
mechanism, as specified in [RFC4585], Section 4.2. The list of
RTCP feedback mechanisms that SHOULD/MUST be supported is
specified in [I-D.ietf-rtcweb-rtp-usage], Section 5.1.
o An "a=ssrc" line, as specified in [RFC5576], Section 4.1,
indicating the SSRC to be used for sending media.
o If RTX is supported for this media type, another "a=ssrc" line
with the RTX SSRC, and an "a=ssrc-group" line, as specified in
[RFC5576], section 4.2, with semantics set to "FID" and including
the primary and RTX SSRCs.
o If FEC is supported for this media type, another "a=ssrc" line
with the FEC SSRC, and an "a=ssrc-group" line, as specified in
[RFC5576], section 4.2, with semantics set to "FEC" and including
the primary and FEC SSRCs.
o [OPEN ISSUE: Handling of a=imageattr]
o [TODO: bundle-only]
Lastly, if a data channel has been created, a m= section MUST be
generated for data. The <media> field MUST be set to "application"
and the <proto> field MUST be set to "DTLS/SCTP", as specified in
[I-D.ietf-mmusic-sctp-sdp], Section 3. The "a=mid", "a=ice-ufrag",
"a=ice-passwd", "a=ice-options", "a=candidate", "a=fingerprint", and
"a=setup" lines MUST be included as mentioned above. [OPEN ISSUE:
additional SCTP-specific stuff to be included, as indicated in
[I-D.jesup-rtcweb-data-protocol] (currently none)]
Once all m= sections have been generated, a session-level "a=group"
attribute MUST be added as specified in [RFC5888]. This attribute
MUST have semantics "BUNDLE", and identify the m= sections to be
bundled. [OPEN ISSUE: Need to determine exactly how this decision is
made.]
Attributes that are common between all m= sections MAY be moved to
session-level, if desired.
Attributes other than the ones specified above MAY be included,
except for the following attributes which are specifically
incompatible with the requirements of [I-D.ietf-rtcweb-rtp-usage],
and MUST NOT be included:
o "a=crypto"
o "a=key-mgmt"
o "a=ice-lite"
Note that when BUNDLE is used, any additional attributes that are
added MUST follow the advice in
[I-D.nandakumar-mmusic-sdp-mux-attributes] on how those attributes
interact with BUNDLE.
5.2.2. Subsequent Offers
When createOffer is called a second (or later) time, the processing
is different, depending on the current signaling state.
If the initial offer was not applied using setLocalDescription,
meaning the PeerConnection is still in the "stable" state, the steps
for generating an initial offer should be followed, with this
exception:
o The "o=" line MUST stay the same.
If the initial offer was applied using setLocalDescription, but an
answer from the remote side has not yet been applied, meaning the
PeerConnection is still in the "local-offer" state, the steps for
generating an initial offer should be followed, with these
exceptions:
o The "o=" line MUST stay the same, except for the <session-version>
field, which MUST increase by 1 from the previously applied local
description.
o The "s=" and "t=" lines MUST stay the same.
o Each "a=mid" line MUST stay the same.
o Each "a=ice-ufrag" and "a=ice-pwd" line MUST stay the same.
o For MediaStreamTracks that are still present, the "a=msid",
"a=ssrc", and "a=ssrc-group" lines MUST stay the same.
o If any MediaStreamTracks have been removed, either through the
removeStream method or by removing them from an added MediaStream,
their m= sections MUST be marked as recvonly by changing the value
of the [RFC3264] directional attribute to "a=recvonly". The
"a=msid", "a=ssrc", and "a=ssrc-group" lines MUST be removed from
the associated m= sections.
If the initial offer was applied using setLocalDescription, and an
answer from the remote side has been applied using
setRemoteDescription, meaning the PeerConnection is in the "remote-
pranswer" or "stable" states, an offer is generated based on the
negotiated session descriptions by following the steps mentioned for
the "local-offer" state above, along with these exceptions: [OPEN
ISSUE: should this be permitted in the remote-pranswer state?]
o If a m= section was rejected, i.e. has had its port set to zero in
either the local or remote description, it MUST remain rejected
and have a zero port in the new offer, as indicated in RFC3264,
Section 5.1.
o If a m= section exists in the current local description, but has
its state set to inactive or recvonly, and a new MediaStreamTrack
is added, the previously existing m= section MUST be recycled
instead of creating a new m= section. [OPEN ISSUE: Nail down
exactly what this means. Should the codecs remain the same?
(No.) Should ICE restart? (No.) Can the "a=mid" attribute be
changed? (Yes?)]
o If a m= section exists in the current local description, but does
not have an associated MediaStreamTrack (i.e. it is inactive or
recvonly), a corresponding m= section MUST be generated in the new
offer, but without "a=msid", "a=ssrc", or "a=ssrc-group"
attributes, and the appropriate directional attribute must be
specified.
In addition, for each previously existing, non-rejected m= section in
the new offer, the following adjustments are made based on the
contents of the corresponding m= section in the current remote
description:
o The m= line and corresponding "a=rtpmap" and "a=fmtp" lines MUST
only include codecs present in the remote description.
o The RTP header extensions MUST only include those that are present
in the remote description.
o The RTCP feedback extensions MUST only include those that are
present in the remote description.
o The "a=rtcp-mux" line MUST only be added if present in the remote
description.
o The "a=rtcp-rsize" line MUST only be added if present in the
remote description.
5.2.3. Constraints Handling
The createOffer method takes as a parameter a MediaConstraints
object. Special processing is performed when generating a SDP
description if the following constraints are present.
5.2.3.1. OfferToReceiveAudio
If the "OfferToReceiveAudio" constraint is specified, with a value of
"true", the offer MUST include a non-rejected m= section with media
type "audio", even if no audio MediaStreamTrack has been added to the
PeerConnection. This allows the offerer to receive audio even when
not sending it; accordingly, the directional attribute on the audio
m= section MUST be set to recvonly. If this constraint is specified
when an audio MediaStreamTrack has already been added to the
PeerConnection, or a non-rejected m= section with media type "audio"
previously existed, it has no effect.
5.2.3.2. OfferToReceiveVideo
If the "OfferToReceiveAudio" constraint is specified, with a value of
"true", the offer MUST include a m= section with media type "video",
even if no video MediaStreamTrack has been added to the
PeerConnection. This allows the offerer to receive video even when
not sending it; accordingly, the directional attribute on the video
m= section MUST be set to recvonly. If this constraint is specified
when an video MediaStreamTrack has already been added to the
PeerConnection, or a non-rejected m= section with media type "video"
previously existed, it has no effect.
5.2.3.3. VoiceActivityDetection
If the "VoiceActivityDetection" constraint is specified, with a value
of "true", the offer MUST indicate support for silence suppression by
including comfort noise ("CN") codecs for each supported clock rate,
as specified in [RFC3389], Section 5.1. [OPEN issue: should this do
anything in signaling, or should it just control built-in DTX modes
in audio codecs? Opus has built-in DTX, but G.711 does not.]
5.2.3.4. IceRestart
If the "IceRestart" constraint is specified, with a value of "true",
the offer MUST indicate an ICE restart by generating new ICE ufrag
and pwd attributes, as specified in RFC5245, Section 9.1.1.1. If
this constraint is specified on an initial offer, it has no effect
(since a new ICE ufrag and pwd are already generated).
5.3. Generating an Answer
When createAnswer is called, a new SDP description must be created
that is compatible with the supplied remote description as well as
the requirements specified in [I-D.ietf-rtcweb-rtp-usage]. The exact
details of this process are explained below.
5.3.1. Initial Answers
When createAnswer is called for the first time after a remote
description has been provided, the result is known as the initial
answer. If no remote description has been installed, an answer
cannot be generated, and an error MUST be returned.
Note that the remote description SDP may not have been created by a
WebRTC endpoint and may not conform to all the requirements listed in
Section 5.2. For many cases, this is not a problem. However, if any
mandatory SDP attributes are missing, or functionality listed as
mandatory-to-use is not present (e.g. ICE, DTLS) [TODO: find
reference for this], this MUST be treated as an error. [OPEN ISSUE:
Should this cause setRemoteDescription to fail, or should this cause
createAnswer to reject those particular m= sections?]
The first step in generating an initial answer is to generate
session-level attributes. The process here is identical to that
indicated in the Initial Offers section above, with the addition that
The next step is to generate m= sections for each m= section that is
present in the remote offer, as specified in [RFC3264], Section 6.
For the purposes of this discussion, any session-level attributes in
the offer that are also valid as media-level attributes SHALL be
considered to be present in each m= section.
If any of the offered m= sections have been rejected, by stopping the
associated remote MediaStreamTrack, the corresponding m= section in
the answer MUST be marked as rejected by setting the port in the m=
line to zero, as indicated in [RFC3264], Section 6., and processing
continues with the next m= section.
For each non-rejected m= section of a given media type, if there is a
local MediaStreamTrack of the specified type which has been added to
the PeerConnection via addStream and not yet associated with a m=
section, the MediaStreamTrack is associated with the m= section at
this time. If there are more m= sections of a certain type than
MediaStreamTracks, some m= sections will not have an associated
MediaStreamTrack. If there are more MediaStreamTracks of a certain
type than m= sections, only the first N MediaStreamTracks will be
able to be associated in the constructed answer. The remainder will
need to be associated in a subsequent offer.
Each m= section should then generated as specified in [RFC3264],
Section 6.1. The <proto> field MUST be set to "RTP/SAVPF". If the
offer supports BUNDLE, all m= sections to be BUNDLEd must use the
same ICE credentials and candidates; all m= sections not being
BUNDLEd must use unique ICE credentials and candidates. Each m=
section MUST include the following:
o If present in the offer, an "a=mid" line, as specified in
[RFC5888], Section 9.1. The "mid" value MUST match that specified
in the offer.
o If a local MediaStreamTrack has been associated, an "a=msid" line,
as specified in [I-D.ietf-mmusic-msid], Section 2.
o [OPEN ISSUE: Use of App Token versus stream-correlator ]
o If a local MediaStreamTrack has been associated, an "a=sendrecv"
line, as specified in [RFC3264], Section 6.1. If no local
MediaStreamTrack has been associated, an "a=recvonly" line.
[TODO: handle non-sendrecv offered m= sections]
o For each supported codec that is present in the offer, "a=rtpmap"
and "a=fmtp" lines, as specified in [RFC4566], Section 6, and
[RFC3264], Section 6.1. For audio, the codecs specified in
[I-D.ietf-rtcweb-audio], Section 3, MUST be be supported. Note
that for simplicity, the answerer MAY use different payload types
for codecs than the offerer, as it is not prohibited by
Section 6.1.
o If "rtx" is present in the offer, for each primary codec where RTP
retransmission should be used, a corresponding "a=rtpmap" line
indicating "rtx" with the clock rate of the primary codec and an
"a=fmtp" line that references the payload type fo the primary
codec, as specified in [RFC4588], Section 8.1.
o For each supported FEC mechanism that is present in the offer, a
corresponding "a=rtpmap" line indicating the desired FEC codec.
o "a=ice-ufrag" and "a=ice-passwd" lines, as specified in [RFC5245],
Section 15.4.
o If the "trickle" ICE option is present in the offer, an "a=ice-
options" line, with the "trickle" option, as specified in
[I-D.ivov-mmusic-trickle-ice], Section 4.
o For each candidate that has been gathered during the most recent
gathering phase, an "a=candidate" line, as specified in [RFC5245],
Section 4.3., paragraph 3.
o For the current default candidate, a "c=" line, as specific in
[RFC5245], Section 4.3., paragraph 6. [OPEN ISSUE, pending
resolution in mmusic: If no candidates have yet been gathered yet,
the default candidate should be set to the null value defined in
[I-D.ivov-mmusic-trickle-ice], Section 5.1.]
o An "a=fingerprint" line, as specified in [RFC4572], Section 5.
Use of the SHA-256 algorithm for the fingerprint is REQUIRED; if
the browser also supports stronger hashes, additional
"a=fingerprint" lines with these hashes MAY also be added.
o An "a=setup" line, as specified in [RFC4145], Section 4, and
clarified for use in DTLS-SRTP scenarios in [RFC5763], Section 5.
The role value in the answer MUST be "active" or "passive"; the
"active" role is RECOMMENDED.
o If present in the offer, an "a=rtcp-mux" line, as specified in
[RFC5761], Section 5.1.1.
o If present in the offer, an "a=rtcp-rsize" line, as specified in
[RFC5506], Section 5.
o For each supported RTP header extension that is present in the
offer, an "a=extmap" line, as specified in [RFC5285], Section 5.
The list of header extensions that SHOULD/MUST be supported is
specified in [I-D.ietf-rtcweb-rtp-usage], Section 5.2. Any header
extensions that require encryption MUST be specified as indicated
in [RFC6904], Section 4.
o For each supported RTCP feedback mechanism that is present in the
offer, an "a=rtcp-fb" mechanism, as specified in [RFC4585],
Section 4.2. The list of RTCP feedback mechanisms that SHOULD/
MUST be supported is specified in [I-D.ietf-rtcweb-rtp-usage],
Section 5.1.
o If a local MediaStreamTrack has been associated, an "a=ssrc" line,
as specified in [RFC5576], Section 4.1, indicating the SSRC to be
used for sending media.
o If a local MediaStreamTrack has been associated, and RTX has been
negotiated for this m= section, another "a=ssrc" line with the RTX
SSRC, and an "a=ssrc-group" line, as specified in [RFC5576],
section 4.2, with semantics set to "FID" and including the primary
and RTX SSRCs.
o If a local MediaStreamTrack has been associated, and FEC has been
negotiated for this m= section, another "a=ssrc" line with the FEC
SSRC, and an "a=ssrc-group" line, as specified in [RFC5576],
section 4.2, with semantics set to "FEC" and including the primary
and FEC SSRCs.
o [OPEN ISSUE: Handling of a=imageattr]
o [TODO: bundle-only]
If a data channel m= section has been offered, a m= section MUST also
be generated for data. The <media> field MUST be set to
"application" and the <proto> field MUST be set to "DTLS/SCTP", as
specified in [I-D.ietf-mmusic-sctp-sdp], Section 3. The "a=mid", "a
=ice-ufrag", "a=ice-passwd", "a=ice-options", "a=candidate",
"a=fingerprint", and "a=setup" lines MUST be included as mentioned
above. [OPEN ISSUE: additional SCTP-specific stuff to be included,
as indicated in [I-D.jesup-rtcweb-data-protocol] (currently none)]
[TODO: processing of BUNDLE group]
Attributes that are common between all m= sections MAY be moved to
session-level, if desired.
The attributes prohibited in creation of offers are also prohibited
in the creation of answers.
5.3.2. Subsequent Answers
5.3.3. Constraints Handling
5.4. Parsing an Offer
5.5. Parsing an Answer
5.6. Applying a Local Description
5.7. Applying a Remote Description
6. Configurable SDP Parameters 6. Configurable SDP Parameters
Note: This section is still very early and is likely to Note: This section is still very early and is likely to significantly
significantly change as we get a better understanding of a) the use change as we get a better understanding of a) the use cases for this
cases for this b) the implications at the protocol level c) feedback b) the implications at the protocol level c) feedback from
from implementors on what they can do. implementors on what they can do.
The following elements of the SDP media description MUST NOT be The following elements of the SDP media description MUST NOT be
changed between the createOffer and the setLocalDescription, since changed between the createOffer and the setLocalDescription, since
they reflect transport attributes that are solely under browser they reflect transport attributes that are solely under browser
control, and the browser MUST NOT honor an attempt to change them: control, and the browser MUST NOT honor an attempt to change them:
o The number, type and port number of m-lines. o The number, type and port number of m-lines.
o The generated ICE credentials (a=ice-ufrag and a=ice-pwd). o The generated ICE credentials (a=ice-ufrag and a=ice-pwd).
o The set of ICE candidates and their parameters (a=candidate). o The set of ICE candidates and their parameters (a=candidate).
The following modifications, if done by the browser to a description The following modifications, if done by the browser to a description
between createOffer/createAnswer and the setLocalDescription, MUST be between createOffer/createAnswer and the setLocalDescription, MUST be
honored by the browser: honored by the browser:
o Remove or reorder codecs (m=) o Remove or reorder codecs (m=)
The following parameters may be controlled by constraints passed into The following parameters may be controlled by constraints passed into
createOffer/createAnswer. As an open issue, these changes may also createOffer/createAnswer. As an open issue, these changes may also
skipping to change at page 22, line 23 skipping to change at page 32, line 19
o Remove or reorder codecs (m=) o Remove or reorder codecs (m=)
The following parameters may be controlled by constraints passed into The following parameters may be controlled by constraints passed into
createOffer/createAnswer. As an open issue, these changes may also createOffer/createAnswer. As an open issue, these changes may also
be be performed by manipulating the SDP returned from createOffer/ be be performed by manipulating the SDP returned from createOffer/
createAnswer, as indicated above, as long as the capabilities of the createAnswer, as indicated above, as long as the capabilities of the
endpoint are not exceeded (e.g. asking for a resolution greater than endpoint are not exceeded (e.g. asking for a resolution greater than
what the endpoint can encode): what the endpoint can encode):
o disable BUNDLE (a=group) o disable BUNDLE (a=group)
o disable RTCP mux (a=rtcp-mux) o disable RTCP mux (a=rtcp-mux)
o change send resolution or framerate
o change desired recv resolution or framerate o change send resolution or frame rate
o change maximum total bandwidth (b=) [OPEN ISSUE: need to clarify
if this is CT or AS - see section 5.8 of RFC4566] o change desired recv resolution or frame rate
o change maximum total bandwidth (b=) [OPEN ISSUE: need to clarify
if this is CT or AS - see section 5.8 of [RFC4566]]
o remove desired AVPF mechanisms (a=rtcp-fb) o remove desired AVPF mechanisms (a=rtcp-fb)
o remove RTP header extensions (a=extmap) o remove RTP header extensions (a=extmap)
o change media send/recv state (a=sendonly/recvonly/inactive) o change media send/recv state (a=sendonly/recvonly/inactive)
For example, an application could implement call hold by adding an For example, an application could implement call hold by adding an
a=inactive attribute to its local description, and then applying and a=inactive attribute to its local description, and then applying and
signaling that description. signaling that description.
The application can also modify the SDP to reduce the capabilities in The application can also modify the SDP to reduce the capabilities in
the offer it sends to the far side in any way the application sees the offer it sends to the far side in any way the application sees
fit, as long as it is a valid SDP offer and specifies a subset of fit, as long as it is a valid SDP offer and specifies a subset of
what the browser is expecting to do. what the browser is expecting to do.
skipping to change at page 23, line 4 skipping to change at page 33, line 8
the other party, and all incoming SDP will be processed by the the other party, and all incoming SDP will be processed by the
browser to the extent of its capabilities. It is an error to assume browser to the extent of its capabilities. It is an error to assume
that all SDP is well-formed; however, one should be able to assume that all SDP is well-formed; however, one should be able to assume
that any implementation of this specification will be able to that any implementation of this specification will be able to
process, as a remote offer or answer, unmodified SDP coming from any process, as a remote offer or answer, unmodified SDP coming from any
other implementation of this specification. other implementation of this specification.
7. Security Considerations 7. Security Considerations
The intent of the WebRTC protocol suite is to provide an environment The intent of the WebRTC protocol suite is to provide an environment
that is securable by default: all media is encrypted, keys are that is securable by default: all media is encrypted, keys are
exchanged in a secure fashion, and the Javascript API includes exchanged in a secure fashion, and the Javascript API includes
functions that can be used to verify the identity of communication functions that can be used to verify the identity of communication
partners. partners.
8. IANA Considerations 8. IANA Considerations
This document requires no actions from IANA. This document requires no actions from IANA.
9. Acknowledgements 9. Acknowledgements
skipping to change at page 23, line 27 skipping to change at page 33, line 31
Neil Stratford, Eric Rescorla, Anant Narayanan, Andrew Hutton, Neil Stratford, Eric Rescorla, Anant Narayanan, Andrew Hutton,
Richard Ejzak, and Adam Bergkvist all provided valuable feedback on Richard Ejzak, and Adam Bergkvist all provided valuable feedback on
this proposal. Matthew Kaufman provided the observation that keeping this proposal. Matthew Kaufman provided the observation that keeping
state out of the browser allows a call to continue even if the page state out of the browser allows a call to continue even if the page
is reloaded. is reloaded.
10. References 10. References
10.1. Normative References 10.1. Normative References
[I-D.rescorla-mmusic-ice-trickle] [I-D.ietf-mmusic-msid]
Rescorla, E., Uberti, J., and E. Ivov, "Trickle ICE: Alvestrand, H., "Cross Session Stream Identification in
Incremental Provisioning of Candidates for the Interactive the Session Description Protocol", draft-ietf-mmusic-
Connectivity Establishment (ICE) Protocol", msid-01 (work in progress), August 2013.
draft-rescorla-mmusic-ice-trickle-00 (work in progress),
October 2012. [I-D.ietf-mmusic-sctp-sdp]
Loreto, S. and G. Camarillo, "Stream Control Transmission
Protocol (SCTP)-Based Media Transport in the Session
Description Protocol (SDP)", draft-ietf-mmusic-sctp-sdp-04
(work in progress), June 2013.
[I-D.ietf-mmusic-sdp-bundle-negotiation]
Holmberg, C., Alvestrand, H., and C. Jennings,
"Multiplexing Negotiation Using Session Description
Protocol (SDP) Port Numbers", draft-ietf-mmusic-sdp-
bundle-negotiation-04 (work in progress), June 2013.
[I-D.ietf-rtcweb-audio]
Valin, J. and C. Bran, "WebRTC Audio Codec and Processing
Requirements", draft-ietf-rtcweb-audio-02 (work in
progress), August 2013.
[I-D.ietf-rtcweb-rtp-usage]
Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time
Communication (WebRTC): Media Transport and Use of RTP",
draft-ietf-rtcweb-rtp-usage-09 (work in progress),
September 2013.
[I-D.nandakumar-mmusic-sdp-mux-attributes]
Nandakumar, S., "A Framework for SDP Attributes when
Multiplexing", draft-nandakumar-mmusic-sdp-mux-
attributes-03 (work in progress), July 2013.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997. Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, [RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
A., Peterson, J., Sparks, R., Handley, M., and E. A., Peterson, J., Sparks, R., Handley, M., and E.
Schooler, "SIP: Session Initiation Protocol", RFC 3261, Schooler, "SIP: Session Initiation Protocol", RFC 3261,
June 2002. June 2002.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model [RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
with Session Description Protocol (SDP)", RFC 3264, with Session Description Protocol (SDP)", RFC 3264, June
June 2002. 2002.
[RFC3388] Camarillo, G., Eriksson, G., Holler, J., and H.
Schulzrinne, "Grouping of Media Lines in the Session
Description Protocol (SDP)", RFC 3388, December 2002.
[RFC3960] Camarillo, G. and H. Schulzrinne, "Early Media and Ringing [RFC4145] Yon, D. and G. Camarillo, "TCP-Based Media Transport in
Tone Generation in the Session Initiation Protocol (SIP)", the Session Description Protocol (SDP)", RFC 4145,
RFC 3960, December 2004. September 2005.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session [RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006. Description Protocol", RFC 4566, July 2006.
[RFC4572] Lennox, J., "Connection-Oriented Media Transport over the
Transport Layer Security (TLS) Protocol in the Session
Description Protocol (SDP)", RFC 4572, July 2006.
[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey, [RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
"Extended RTP Profile for Real-time Transport Control "Extended RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July
July 2006. 2006.
[RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for [RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for
Real-time Transport Control Protocol (RTCP)-Based Feedback Real-time Transport Control Protocol (RTCP)-Based Feedback
(RTP/SAVPF)", RFC 5124, February 2008. (RTP/SAVPF)", RFC 5124, February 2008.
[RFC5245] Rosenberg, J., "Interactive Connectivity Establishment [RFC5245] Rosenberg, J., "Interactive Connectivity Establishment
(ICE): A Protocol for Network Address Translator (NAT) (ICE): A Protocol for Network Address Translator (NAT)
Traversal for Offer/Answer Protocols", RFC 5245, Traversal for Offer/Answer Protocols", RFC 5245, April
April 2010. 2010.
[RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP
Header Extensions", RFC 5285, July 2008.
[RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and [RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
Control Packets on a Single Port", RFC 5761, April 2010. Control Packets on a Single Port", RFC 5761, April 2010.
[RFC5888] Camarillo, G. and H. Schulzrinne, "The Session Description [RFC5888] Camarillo, G. and H. Schulzrinne, "The Session Description
Protocol (SDP) Grouping Framework", RFC 5888, June 2010. Protocol (SDP) Grouping Framework", RFC 5888, June 2010.
10.2. Informative References [RFC6904] Lennox, J., "Encryption of Header Extensions in the Secure
Real-time Transport Protocol (SRTP)", RFC 6904, April
[I-D.alvestrand-mmusic-msid] 2013.
Alvestrand, H., "Cross Session Stream Identification in
the Session Description Protocol",
draft-alvestrand-mmusic-msid-01 (work in progress),
October 2012.
[I-D.holmberg-mmusic-sdp-bundle-negotiation] 10.2. Informative References
Holmberg, C. and H. Alvestrand, "Multiplexing Negotiation
Using Session Description Protocol (SDP) Port Numbers",
draft-holmberg-mmusic-sdp-bundle-negotiation-00 (work in
progress), October 2011.
[I-D.ietf-rtcweb-rtp-usage] [I-D.ivov-mmusic-trickle-ice]
Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time Ivov, E., Rescorla, E., and J. Uberti, "Trickle ICE:
Communication (WebRTC): Media Transport and Use of RTP", Incremental Provisioning of Candidates for the Interactive
draft-ietf-rtcweb-rtp-usage-04 (work in progress), Connectivity Establishment (ICE) Protocol", draft-ivov-
July 2012. mmusic-trickle-ice-01 (work in progress), March 2013.
[I-D.jennings-rtcweb-signaling] [I-D.jennings-rtcweb-signaling]
Jennings, C., Rosenberg, J., and R. Jesup, "RTCWeb Offer/ Jennings, C., Rosenberg, J., and R. Jesup, "RTCWeb Offer/
Answer Protocol (ROAP)", Answer Protocol (ROAP)", draft-jennings-rtcweb-
draft-jennings-rtcweb-signaling-01 (work in progress), signaling-01 (work in progress), October 2011.
October 2011.
[I-D.jesup-rtcweb-data-protocol]
Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel
Protocol", draft-jesup-rtcweb-data-protocol-04 (work in
progress), February 2013.
[I-D.nandakumar-rtcweb-sdp] [I-D.nandakumar-rtcweb-sdp]
Nandakumar, S. and C. Jennings, "SDP for the WebRTC", Nandakumar, S. and C. Jennings, "SDP for the WebRTC",
draft-nandakumar-rtcweb-sdp-00 (work in progress), draft-nandakumar-rtcweb-sdp-02 (work in progress), July
October 2012. 2013.
[RFC3389] Zopf, R., "Real-time Transport Protocol (RTP) Payload for
Comfort Noise (CN)", RFC 3389, September 2002.
[RFC3556] Casner, S., "Session Description Protocol (SDP) Bandwidth [RFC3556] Casner, S., "Session Description Protocol (SDP) Bandwidth
Modifiers for RTP Control Protocol (RTCP) Bandwidth", Modifiers for RTP Control Protocol (RTCP) Bandwidth", RFC
RFC 3556, July 2003. 3556, July 2003.
[RFC3960] Camarillo, G. and H. Schulzrinne, "Early Media and Ringing
Tone Generation in the Session Initiation Protocol (SIP)",
RFC 3960, December 2004.
[RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session [RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session
Description Protocol (SDP) Security Descriptions for Media Description Protocol (SDP) Security Descriptions for Media
Streams", RFC 4568, July 2006. Streams", RFC 4568, July 2006.
[RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
July 2006.
[RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size [RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size
Real-Time Transport Control Protocol (RTCP): Opportunities Real-Time Transport Control Protocol (RTCP): Opportunities
and Consequences", RFC 5506, April 2009. and Consequences", RFC 5506, April 2009.
[RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific [RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific
Media Attributes in the Session Description Protocol Media Attributes in the Session Description Protocol
(SDP)", RFC 5576, June 2009. (SDP)", RFC 5576, June 2009.
[RFC5763] Fischl, J., Tschofenig, H., and E. Rescorla, "Framework [RFC5763] Fischl, J., Tschofenig, H., and E. Rescorla, "Framework
for Establishing a Secure Real-time Transport Protocol for Establishing a Secure Real-time Transport Protocol
(SRTP) Security Context Using Datagram Transport Layer (SRTP) Security Context Using Datagram Transport Layer
Security (DTLS)", RFC 5763, May 2010. Security (DTLS) ", RFC 5763, May 2010.
[W3C.WD-webrtc-20111027] [W3C.WD-webrtc-20111027]
Bergkvist, A., Burnett, D., Narayanan, A., and C. Bergkvist, A., Burnett, D., Narayanan, A., and C.
Jennings, "WebRTC 1.0: Real-time Communication Between Jennings, "WebRTC 1.0: Real-time Communication Between
Browsers", World Wide Web Consortium WD WD-webrtc- Browsers", World Wide Web Consortium WD WD-
20111027, October 2011, webrtc-20111027, October 2011,
<http://www.w3.org/TR/2011/WD-webrtc-20111027>. <http://www.w3.org/TR/2011/WD-webrtc-20111027>.
Appendix A. JSEP Implementation Examples Appendix A. JSEP Implementation Examples
A.1. Example API Flows A.1. Example API Flows
Below are several sample flows for the new PeerConnection and library Below are several sample flows for the new PeerConnection and library
APIs, demonstrating when the various APIs are called in different APIs, demonstrating when the various APIs are called in different
situations and with various transport protocols. For clarity and situations and with various transport protocols. For clarity and
simplicity, the createOffer/createAnswer calls are assumed to be simplicity, the createOffer/createAnswer calls are assumed to be
skipping to change at page 31, line 27 skipping to change at page 41, line 24
OffererUA->OffererJS: onaddstream(remoteStream); OffererUA->OffererJS: onaddstream(remoteStream);
// ICE Completes (at Offerer) // ICE Completes (at Offerer)
OffererUA->AnswererUA: Media OffererUA->AnswererUA: Media
// 200 OK arrives at Offerer // 200 OK arrives at Offerer
OffererJS: answer = parseResponse(sip); OffererJS: answer = parseResponse(sip);
OffererJS->OffererUA: pc.setRemoteDescription("answer", answer); OffererJS->OffererUA: pc.setRemoteDescription("answer", answer);
OffererJS->AnswererJS: ACK OffererJS->AnswererJS: ACK
A.2. Example Session Descriptions
A.2.1. createOffer
This SDP shows a typical initial offer, created by createOffer for a
PeerConnection with a single audio MediaStreamTrack, a single video
MediaStreamTrack, and a single data channel. Host candidates have
also already been gathered. Note some lines have been broken into
two lines for formatting reasons.
v=0
o=- 4962303333179871722 1 IN IP4 0.0.0.0
s=-
t=0 0
a=group:BUNDLE audio video data
m=audio 56500 RTP/SAVPF 111 0 8 126
c=IN IP4 192.0.2.1
a=rtcp:56501 IN IP4 192.0.2.1
a=candidate:3348148302 1 udp 2113937151 192.0.2.1 56500
typ host generation 0
a=candidate:3348148302 2 udp 2113937151 192.0.2.1 56501
typ host generation 0
a=ice-ufrag:ETEn1v9DoTMB9J4r
a=ice-pwd:OtSK0WpNtpUjkY4+86js7ZQl
a=ice-options:trickle
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=rtcp-rsize
a=fingerprint:sha-256
19:E2:1C:3B:4B:9F:81:E6:B8:5C:F4:A5:A8:D8:73:04
:BB:05:2F:70:9F:04:A9:0E:05:E9:26:33:E8:70:88:A2
a=setup:actpass
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:1732846380 cname:EocUG1f0fcg/yvY7
a=msid:47017fee-b6c1-4162-929c-a25110252400
f83006c5-a0ff-4e0a-9ed9-d3e6747be7d9
m=video 56502 RTP/SAVPF 100 115 116 117
c=IN IP4 192.0.2.1
a=rtcp:56503 IN IP4 192.0.2.1
a=candidate:3348148302 1 udp 2113937151 192.0.2.1 56502
typ host generation 0
a=candidate:3348148302 2 udp 2113937151 192.0.2.1 56503
typ host generation 0
a=ice-ufrag:BGKkWnG5GmiUpdIV
a=ice-pwd:mqyWsAjvtKwTGnvhPztQ9mIf
a=ice-options:trickle
a=mid:video
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=rtcp-mux
a=rtcp-rsize
a=fingerprint:sha-256
19:E2:1C:3B:4B:9F:81:E6:B8:5C:F4:A5:A8:D8:73:04
:BB:05:2F:70:9F:04:A9:0E:05:E9:26:33:E8:70:88:A2
a=setup:actpass
a=rtpmap:100 VP8/90000
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 goog-remb
a=rtpmap:115 rtx/90000
a=fmtp:115 apt=100
a=rtpmap:116 red/90000
a=rtpmap:117 ulpfec/90000
a=ssrc:1366781083 cname:EocUG1f0fcg/yvY7
a=ssrc:1366781084 cname:EocUG1f0fcg/yvY7
a=ssrc:1366781085 cname:EocUG1f0fcg/yvY7
a=ssrc-group:FID 1366781083 1366781084
a=ssrc-group:FEC 1366781083 1366781085
a=msid:61317484-2ed4-49d7-9eb7-1414322a7aae
f30bdb4a-5db8-49b5-bcdc-e0c9a23172e0
m=application 56504 DTLS/SCTP 5000
c=IN IP4 192.0.2.1
a=candidate:3348148302 1 udp 2113937151 192.0.2.1 56504
typ host generation 0
a=ice-ufrag:VD5v2BnbZm3mgP3d
a=ice-pwd:+Jlkuox+VVIUDqxcfIDuTZMH
a=ice-options:trickle
a=mid:data
a=fingerprint:sha-256 19:E2:1C:3B:4B:9F:81:E6:B8:5C:F4:A5:A8:D8:73:04
:BB:05:2F:70:9F:04:A9:0E:05:E9:26:33:E8:70:88:A2
a=setup:actpass
a=fmtp:5000 protocol=webrtc-datachannel; streams=10
A.2.2. createAnswer
This SDP shows a typical initial answer to the above offer, created
by createAnswer for a PeerConnection with a single audio
MediaStreamTrack, a single video MediaStreamTrack, and a single data
channel. Host candidates have also already been gathered. Note some
lines have been broken into two lines for formatting reasons.
v=0
o=- 6729291447651054566 1 IN IP4 0.0.0.0
s=-
t=0 0
a=group:BUNDLE audio video data
m=audio 20000 RTP/SAVPF 111 0 8 126
c=IN IP4 192.0.2.2
a=candidate:2299743422 1 udp 2113937151 192.0.2.2 20000
typ host generation 0
a=ice-ufrag:6sFvz2gdLkEwjZEr
a=ice-pwd:cOTZKZNVlO9RSGsEGM63JXT2
a=fingerprint:sha-256 6B:8B:F0:65:5F:78:E2:51:3B:AC:6F:F3:3F:46:1B:35
:DC:B8:5F:64:1A:24:C2:43:F0:A1:58:D0:A1:2C:19:08
a=setup:active
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:3429951804 cname:Q/NWs1ao1HmN4Xa5
a=msid:PI39StLS8W7ZbQl1sJsWUXkr3Zf12fJUvzQ1
PI39StLS8W7ZbQl1sJsWUXkr3Zf12fJUvzQ1a0
m=video 20000 RTP/SAVPF 100 115 116 117
c=IN IP4 192.0.2.2
a=candidate:2299743422 1 udp 2113937151 192.0.2.2 20000
typ host generation 0
a=ice-ufrag:6sFvz2gdLkEwjZEr
a=ice-pwd:cOTZKZNVlO9RSGsEGM63JXT2
a=fingerprint:sha-256 6B:8B:F0:65:5F:78:E2:51:3B:AC:6F:F3:3F:46:1B:35
:DC:B8:5F:64:1A:24:C2:43:F0:A1:58:D0:A1:2C:19:08
a=setup:active
a=mid:video
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=rtcp-mux
a=rtpmap:100 VP8/90000
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 goog-remb
a=rtpmap:115 rtx/90000
a=fmtp:115 apt=100
a=rtpmap:116 red/90000
a=rtpmap:117 ulpfec/90000
a=ssrc:3229706345 cname:Q/NWs1ao1HmN4Xa5
a=ssrc:3229706346 cname:Q/NWs1ao1HmN4Xa5
a=ssrc:3229706347 cname:Q/NWs1ao1HmN4Xa5
a=ssrc-group:FID 3229706345 3229706346
a=ssrc-group:FEC 3229706345 3229706347
a=msid:PI39StLS8W7ZbQl1sJsWUXkr3Zf12fJUvzQ1
PI39StLS8W7ZbQl1sJsWUXkr3Zf12fJUvzQ1v0
m=application 20000 DTLS/SCTP 5000
c=IN IP4 192.0.2.2
a=candidate:2299743422 1 udp 2113937151 192.0.2.2 20000
typ host generation 0
a=ice-ufrag:6sFvz2gdLkEwjZEr
a=ice-pwd:cOTZKZNVlO9RSGsEGM63JXT2
a=fingerprint:sha-256 6B:8B:F0:65:5F:78:E2:51:3B:AC:6F:F3:3F:46:1B:35
:DC:B8:5F:64:1A:24:C2:43:F0:A1:58:D0:A1:2C:19:08
a=setup:active
a=mid:data
a=fmtp:5000 protocol=webrtc-datachannel; streams=10
Appendix B. Change log Appendix B. Change log
Changes in draft-04:
o Filled in sections on createOffer and createAnswer.
o Added SDP examples.
o Fixed references.
Changes in draft-03: Changes in draft-03:
o Added text describing relationship to W3C specification o Added text describing relationship to W3C specification
Changes in draft -02: Changes in draft-02:
o Converted from nroff o Converted from nroff
o Removed comparisons to old approaches abandoned by the working o Removed comparisons to old approaches abandoned by the working
group group
o Removed stuff that has moved to W3C specificaiton
o Removed stuff that has moved to W3C specification
o Align SDP handling with W3C draft o Align SDP handling with W3C draft
o Clarified section on forking. o Clarified section on forking.
Changes in draft -01: Changes in draft-01:
o Added diagrams for architecture and state machine. o Added diagrams for architecture and state machine.
o Added sections on forking and rehydration. o Added sections on forking and rehydration.
o Clarified meaning of "pranswer" and "answer". o Clarified meaning of "pranswer" and "answer".
o Reworked how ICE restarts and media directions are controlled. o Reworked how ICE restarts and media directions are controlled.
o Added list of parameters that can be changed in a description. o Added list of parameters that can be changed in a description.
o Updated suggested API and examples to match latest thinking. o Updated suggested API and examples to match latest thinking.
o Suggested API and examples have been moved to an appendix. o Suggested API and examples have been moved to an appendix.
Changes in draft -00: Changes in draft -00:
o Migrated from draft-uberti-rtcweb-jsep-02. o Migrated from draft-uberti-rtcweb-jsep-02.
Authors' Addresses Authors' Addresses
Justin Uberti Justin Uberti
Google Google
747 6th Ave S 747 6th Ave S
Kirkland, WA 98033 Kirkland, WA 98033
USA USA
Email: justin@uberti.name Email: justin@uberti.name
Cullen Jennings Cullen Jennings
Cisco Cisco
170 West Tasman Drive 170 West Tasman Drive
San Jose, CA 95134 San Jose, CA 95134
USA USA
Email: fluffy@iii.ca Email: fluffy@iii.ca
 End of changes. 82 change blocks. 
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