[asterisk-dev] [Code Review] Force directmedia sessions to use a common codec

[David Vossel]

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Is there anyway to disable direct media if we detect there are no common codecs available on the re-invite?

- David


On 2010-10-06 00:04:42, Terry Wilson wrote:
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> (Updated 2010-10-06 00:04:42)
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> Review request for Asterisk Developers.
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> Summary
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> From the issue:
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> It is "well known" that setting up directmedia paths between SIP devices with different preferred-codec lists can be problematic. The solution has previously been to either disable directmedia, or use the codec-negotiation patch which is available for Asterisk 1.2 and 1.4.
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> PROBLEM: In its simplest form, the problem is (I believe) as follows:
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> When RTP tries to set up a direct media path, it does so by setting each party separately, and giving each party free reign over their codec choice. Thus under some circumstances the 2 remote devices can make 2 choices which are different. If the 2 parties disagree on the negotiated codec, you generally get silence in both directions.
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> SOLUTION: Make rtp.c just a bit more clever about what codecs each party is offered - In fact, do not allow a choice, just tell them what is acceptable.
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> EXAMPLE: Device A (g722|alaw) calls via asterisk to device B (alaw) - Initially a transcoded path is setup between the 2 parties. directmedia is enabled, and they have a common codec (alaw).
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> At present, during negotiation, RTP tells A to use (alaw) and B to use (g722|alaw) - chan_sip.c then notes those codec requests, ignores them and tells A to use (g722|alaw) and B to use (alaw)
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> Of course this results in A using g722, and B using alaw - Resulting in silence.
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> The patch (to be attached) changes:
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> main/rtp.c to make it choose a single audio and video codec where possible before passing the directmedia request on to the channels.
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> channels/chan_sip.c to make it use the passed in codec offering from RTP as long as it is doing a directmedia reinvite - At any other time, old behaviour remains, allowing better codec choices where possible.
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> This addresses bug 17403.
>     https://issues.asterisk.org/view.php?id=17403
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> Diffs
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>   /trunk/channels/chan_sip.c 290542 
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> Diff: https://reviewboard.asterisk.org/r/967/diff
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> Testing
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> I verified the issue by recreating the aforementioned scenario. I ported the patch to trunk and verified that audio was negotiated properly for that scenario.
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> Thanks,
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> Terry
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>