[asterisk-dev] [Code Review] Send the SRCCHANGE indication when we do a masquerade

[Jeff Peeler]

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Ship it!


So I guess you decided that it was unnecessary to change the indications in ast_channel_bridge? I didn't look any further than knowing that after a masquerade they were sent out, which I thought perhaps might be enough rather than adding indications in the masquerade code. If not, then cool.

- Jeff


On 2010-08-13 14:14:30, Terry Wilson wrote:
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> (Updated 2010-08-13 14:14:30)
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> Review request for Asterisk Developers and Jeff Peeler.
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> Summary
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> From the issue:
> On every SIP Transfer (Example: A calls B / B places A on hold / B calls C / A sends Transfer to Asterisk PBX) the Outing RTP Traffic from Asterisk to the transfer target (RTP to C) is broken. The Asterisk is changing the RTP Timestamp massively but the SSRC stays on the old value and the timestamp marker is also not set. As soon as the new timestamp is smaller than the old timestamp value the transfer target rejects the RTP Packets after the transfer (Not really, it's just not played), so i get one way audio.
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> This patch sends the SRCCHANGE indication so that the RTP SSRC will change for the original and bridged channels (if they use RTP for media).
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> This addresses bug 17007.
>     https://issues.asterisk.org/view.php?id=17007
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> Diffs
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>   /branches/1.4/main/channel.c 282202 
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> Diff: https://reviewboard.asterisk.org/r/862/diff
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> Testing
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> End-user reports that a similar patch worked. I have made some test calls involving transfer and verified that the SSRC changed (when we aren't P2P bridging).
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> Thanks,
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> Terry
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>