--- 1/draft-ietf-rtcweb-audio-04.txt	2014-02-13 18:14:39.314904733 -0800
+++ 2/draft-ietf-rtcweb-audio-05.txt	2014-02-13 18:14:39.330905136 -0800
@@ -1,19 +1,19 @@
 
 Network Working Group                                          JM. Valin
 Internet-Draft                                                   Mozilla
 Intended status: Standards Track                                 C. Bran
-Expires: July 31, 2014                                       Plantronics
-                                                        January 27, 2014
+Expires: August 17, 2014                                     Plantronics
+                                                       February 13, 2014
 
              WebRTC Audio Codec and Processing Requirements
-                       draft-ietf-rtcweb-audio-04
+                       draft-ietf-rtcweb-audio-05
 
 Abstract
 
    This document outlines the audio codec and processing requirements
    for WebRTC client application and endpoint devices.
 
 Status of This Memo
 
    This Internet-Draft is submitted in full conformance with the
    provisions of BCP 78 and BCP 79.
@@ -21,21 +21,21 @@
    Internet-Drafts are working documents of the Internet Engineering
    Task Force (IETF).  Note that other groups may also distribute
    working documents as Internet-Drafts.  The list of current Internet-
    Drafts is at http://datatracker.ietf.org/drafts/current/.
 
    Internet-Drafts are draft documents valid for a maximum of six months
    and may be updated, replaced, or obsoleted by other documents at any
    time.  It is inappropriate to use Internet-Drafts as reference
    material or to cite them other than as "work in progress."
 
-   This Internet-Draft will expire on July 31, 2014.
+   This Internet-Draft will expire on August 17, 2014.
 
 Copyright Notice
 
    Copyright (c) 2014 IETF Trust and the persons identified as the
    document authors.  All rights reserved.
 
    This document is subject to BCP 78 and the IETF Trust's Legal
    Provisions Relating to IETF Documents
    (http://trustee.ietf.org/license-info) in effect on the date of
    publication of this document.  Please review these documents
@@ -74,27 +74,26 @@
 
 3.  Codec Requirements
 
    To ensure a baseline level of interoperability between WebRTC
    clients, a minimum set of required codecs are specified below.  If
    other suitable audio codecs are available for the browser to use, it
    is RECOMMENDED that they are also be included in the offer in order
    to maximize the possibility to establish the session without the need
    for audio transcoding.
 
-   WebRTC clients are REQUIRED to implement the following audio codecs.
+   WebRTC clients are REQUIRED to implement the following audio codecs:
 
-   o  Opus [RFC6716], with the payload format specified in [Opus-RTP]
-      and any ptime value up to 120 ms
+   o  Opus [RFC6716] with the payload format specified in [Opus-RTP].
 
-   o  G.711 PCMA and PCMU with one channel, a rate of 8000 Hz and any
-      ptime value up to 120 ms - see section 4.5.14 of [RFC3551]
+   o  G.711 PCMA and PCMU with the payload format specified in section
+      4.5.14 of [RFC3551].
 
    o  The audio/telephone-event media format as specified in [RFC4733].
       WebRTC clients are REQUIRED to be able to generate and consume the
       following events:
 
       +------------+--------------------------------+-----------+
       |Event Code  | Event Name                     | Reference |
       +------------+--------------------------------+-----------+
       | 0          | DTMF digit "0"                 |  RFC4733  |
       | 1          | DTMF digit "1"                 |  RFC4733  |