[asterisk-dev] [Code Review] 4378: bridge / res_pjsip_sdp_rtp: Fix issues with media not being reinvited during direct media.

[Joshua Colp]

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(Updated Jan. 27, 2015, 4:49 p.m.)


Review request for Asterisk Developers.


Changes
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Changed approach to stopping dissolve per Richard.


Repository: Asterisk


Description
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Currently there exists two issues which prevent direct media from being reinvited depending on the scenario:

1. During a swap operation for a brief period of time there will exist 3 channels in a bridge. This is NOT handled by the bridge_native_rtp module and causes it to not reinvite one of the channels that it should when it may be leaving. As it's a reasonable expectation for a bridge technology which can only handle 2 channels to only ever see 2 I've moved the operation which causes the swap channel to leave to before the new channel is actually added to the bridge. This means bridge_native_rtp only sees the two channels it saw previously and reinvites occur as expected.

2. If the res_pjsip_sdp_rtp module received a re-invite *AFTER* the session had been established it did not notify upstream that things such as the bridge_native_rtp module should re-evaluate and potentially reinvite the remote side. The res_pjsip_sdp_rtp module will now do this using the UPDATE_RTP_PEER control frame if an offer is received after the session is established.


Diffs (updated)
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  /branches/13/res/res_pjsip_sdp_rtp.c 431113 
  /branches/13/main/bridge_channel.c 431113 

Diff: https://reviewboard.asterisk.org/r/4378/diff/


Testing
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Tried various scenarios including attended transfers and multiple Asterisk instances in the path. Previously media would go via the wrong route or not at all. With patch reinvites occur as expected.


Thanks,

Joshua Colp

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