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

[Joshua Colp]

> On Jan. 27, 2015, 3:57 p.m., rmudgett wrote:
> > /branches/13/main/bridge_channel.c, lines 1998-2001
> > <https://reviewboard.asterisk.org/r/4378/diff/1/?file=71097#file71097line1998>
> >
> >     The reason a swap channel is pulled after the new channel is pushed is because pulling the last channel out of a bridge may dissolve it!  This is not good.  If the AST_BRIDGE_FLAG_DISSOLVE_EMPTY flag is set the bridge will dissolve.  Most mixing bridges are going to have this flag set.
> >     
> >     The bridge technology must deal only with the channels that the bridge core has told it about with the technology join/leave callbacks.  Otherwise, what is the point of the join/leave technology callbacks if the technology is going to look at the bridge channel member list at inappropriate times?
> 
> Joshua Colp wrote:
>     That flag can be unset and reset appropriately. The bridge is locked during this operation.
>     
>     I firmly disagree with your statement about the bridge technology only dealing with channels it has been told about. That means each bridge technology has to DUPLICATE information that the core already has. And the point of the join/leave callbacks is so the bridge technology can react to the act of a channel joining or leaving a bridge. In case of bridge_native_rtp that means setting up local bridging or doing reinvites.

Additionally: During testing there was always one other channel in the bridge so the dissolve check never kicked in.


- Joshua


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On Jan. 27, 2015, 12:06 p.m., Joshua Colp wrote:
> 
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> 
> (Updated Jan. 27, 2015, 12:06 p.m.)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Repository: Asterisk
> 
> 
> Description
> -------
> 
> 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
> -----
> 
>   /branches/13/res/res_pjsip_sdp_rtp.c 431112 
>   /branches/13/main/bridge_channel.c 431112 
> 
> Diff: https://reviewboard.asterisk.org/r/4378/diff/
> 
> 
> Testing
> -------
> 
> 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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