Review Board 1.7.16


Pimp My SIP Media Improvements

Review Request #2318 - Created Feb. 6, 2013 and submitted

Joshua Colp
/team/file/pimp_sip_media
Reviewers
asterisk-dev
Asterisk
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.
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)
Review request changed
Updated (Feb. 12, 2013, 2:31 a.m.)
Fixed a bug with SDP negotiation and also made it so RTP automatically chooses IPv6 or IPv4 for incoming sessions.
Ship it!
Posted (Feb. 12, 2013, 11:53 a.m.)
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!

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