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

[Russell Bryant]

-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/862/#review2602
-----------------------------------------------------------

Ship it!


- Russell


On 2010-08-13 14:14:30, Terry Wilson wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/862/
> -----------------------------------------------------------
> 
> (Updated 2010-08-13 14:14:30)
> 
> 
> Review request for Asterisk Developers and Jeff Peeler.
> 
> 
> Summary
> -------
> 
> 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.
> 
> 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).
> 
> 
> This addresses bug 17007.
>     https://issues.asterisk.org/view.php?id=17007
> 
> 
> Diffs
> -----
> 
>   /branches/1.4/main/channel.c 282202 
> 
> Diff: https://reviewboard.asterisk.org/r/862/diff
> 
> 
> Testing
> -------
> 
> 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).
> 
> 
> Thanks,
> 
> Terry
> 
>