[asterisk-dev] [Code Review] Pimp My SIP Media Improvements

[Mark Michelson]

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Ship it!


The only objection I have to this is that some of the SDP handler callbacks in res_sip_sdp_audio are a bit long. But given that I think cleanup and such things are on the horizon, I'm more concerned with functionality in this review. So, ship it!

- Mark


On Feb. 12, 2013, 8:31 a.m., Joshua Colp wrote:
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> https://reviewboard.asterisk.org/r/2318/
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> (Updated Feb. 12, 2013, 8:31 a.m.)
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> 
> Review request for Asterisk Developers.
> 
> 
> Summary
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> 
> These changes clean up media handling, move some more stuff into res_sip_sdp_audio, fixes a few bugs, and adds some additional features.
> 
> The act of negotiating an SDP media stream and actually applying the media stream are now separate operations.
> Hold/unhold works.
> RTP over IPv6 works.
> Use of the 'ptime' attribute works.
> Local Packet2Packet bridging works.
> Symmetric RTP can now be enabled per-endpoint.
> Reduced memory pool usage.
> Fixed bug where the RTP instance was never destroyed.
> 
> 
> Diffs
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> 
>   /team/group/pimp_my_sip/channels/chan_gulp.c 381276 
>   /team/group/pimp_my_sip/configs/res_sip.conf.sample 381276 
>   /team/group/pimp_my_sip/include/asterisk/res_sip.h 381276 
>   /team/group/pimp_my_sip/include/asterisk/res_sip_session.h 381276 
>   /team/group/pimp_my_sip/res/res_sip/sip_configuration.c 381276 
>   /team/group/pimp_my_sip/res/res_sip_sdp_audio.c 381276 
>   /team/group/pimp_my_sip/res/res_sip_session.c 381276 
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> Diff: https://reviewboard.asterisk.org/r/2318/diff
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> 
> Testing
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> 1. Sent and received calls from a few different devices
> 2. Held/unheld a call
> 3. Attempted to set up incompatible calls (only configured for gsm, but offering ulaw only)
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> 
> Thanks,
> 
> Joshua
> 
>

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