Attachment 80243 Details for Bug 65311 – Prefer dynamic PT 101 for telephone-event at clock rate 8000

[patch] Prefer dynamic PT 101 for telephone-event at clock rate 8000

0001-Prefer-dynamic-PT-101-for-telephone-event-at-clock-r.patch (text/plain), 2.17 KB, created by Simon McVittie on 2013-06-03 19:08:19 UTC

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Creator: Simon McVittie

Created: 2013-06-03 19:08:19 UTC

Size: 2.17 KB

patch

obsolete

>From 3d343abe5afe9930f51405785f22a440bde969a8 Mon Sep 17 00:00:00 2001
>From: Simon McVittie <simon.mcvittie@collabora.co.uk>
>Date: Mon, 3 Jun 2013 20:06:53 +0100
>Subject: [PATCH] Prefer dynamic PT 101 for telephone-event at clock rate 8000
>
>The WebRTC implementation in Google Chrome <= 26 would reject calls
>if there was "a telephone-event payload type less than 101"[1] and
>as of 2013-06-03, the Google Mail web UI with the VoIP extension
>seems to have a similar signalling bug. Experimenting with the web UI
>indicates that telephone-events with clock rate != 8000 are irrelevant,
>and only clock rate 8000 matters. Hopefully the same was true in WebRTC
>(I can't find a libjingle commit that looks likely to have fixed this).
>
>Meanwhile, many SIP implementations and at least one Jingle
>implementation (Freeswitch's mod_dingaling) either hard-code payload
>type 101 to be telephone-event, or make the payload type for
>telephone-event a configuration option. I can't help thinking this
>was not how dynamic payload types were meant to work, but
>interoperability is interoperability...
>
>This fixes interop when Empathy 3.8 + telepathy-gabble 0.17.4, on a
>system with not many codecs installed) calls the Google Mail web
>UI. When the same setup is called by a peer that specifies a different
>PT for telephone-event:8000 (the Google Mail web UI uses 126 in its
>outgoing calls), the peer's choice of PT takes precedence.
>
>[1] https://code.google.com/p/webrtc/issues/detail?id=1783
>---
> gst/fsrtpconference/default-codec-preferences |    6 ++++++
> 1 file changed, 6 insertions(+)
>
>diff --git a/gst/fsrtpconference/default-codec-preferences b/gst/fsrtpconference/default-codec-preferences
>index afe737e..e7ceb8a 100644
>--- a/gst/fsrtpconference/default-codec-preferences
>+++ b/gst/fsrtpconference/default-codec-preferences
>@@ -18,6 +18,12 @@ clock-rate=16000
> 
> [audio/ILBC]
> 
>+# It seems to be pretty common to hard-code that dynamic payload type 101
>+# is telephone-event at clock rate 8000. This also avoids an interop
>+# bug with Google Chrome's WebRTC, GMail and possibly other implementations.
>+[audio/telephone-event:8000]
>+clock-rate=8000
>+id=101
> 
> # Disabled audio codecs
> # These are unsuitable for calling
>-- 
>1.7.10.4
>

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