[asterisk-dev] [Code Review] Fix SRTP for changing SSRC and multiple a=crypto SDP lines

[Bryant Zimmerman]

Could this also be causing some timing issues with in-call DTMF? The issues 
did not appear to be in 1.6.2.11 but have showed up in 1.6.2.12-rc1?

Bryant

----------------------------------------
 From: "Russell Bryant" <russell at digium.com>
Sent: Wednesday, September 01, 2010 2:41 PM
To: "Asterisk Developers" <asterisk-dev at lists.digium.com>, "Russell Bryant" 
<russell at digium.com>, "Terry Wilson" <twilson at digium.com>
Subject: Re: [asterisk-dev] [Code Review] Fix SRTP for changing SSRC and 
multiple a=crypto SDP lines

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This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/878/#review2651
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Ship it!

- Russell

On 2010-08-26 01:29:16, Terry Wilson wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/878/
> -----------------------------------------------------------
> 
> (Updated 2010-08-26 01:29:16)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Summary
> -------
> 
> Adding code to Asterisk that changed the SSRC during bridges and 
masquerades broke SRTP functionality. Also broken was handling the 
situation where an incoming INVITE had more than one crypto offer. This 
patch caches the SRTP policies the we use so that we can change the ssrc 
and inform libsrtp of the new streams. It also uses the first acceptable 
a=crypto line from the incoming INVITE.
> 
> 
> This addresses bug 17563.
> https://issues.asterisk.org/view.php?id=17563
> 
> 
> Diffs
> -----
> 
> /branches/1.8/channels/chan_sip.c 283320 
> /branches/1.8/include/asterisk/res_srtp.h 283320 
> /branches/1.8/main/rtp_engine.c 283320 
> /branches/1.8/res/res_rtp_asterisk.c 283320 
> /branches/1.8/res/res_srtp.c 283320 
> 
> Diff: https://reviewboard.asterisk.org/r/878/diff
> 
> 
> Testing
> -------
> 
> I tested by 1) Setting up Polycom phones to send two a=crypto lines 2) 
Changing SIP hold/unhold to call the rtp change_source callback to verify 
that changing source worked 3) Doing transfers that would cause a 
masquerade and therefore a source change 4) astobj2 show stats to verify 
that there were no object leaks with the above tests.
> 
> 
> Thanks,
> 
> Terry
> 
>

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