|Summary:||ASTERISK-18971: Inbound Gtalk calls fail randomly|
|Reporter:||Cabel McCoy (cabelmccoy)||Labels:|
|Date Opened:||2011-12-05 22:47:33.000-0600||Date Closed:||2015-02-26 09:37:29.000-0600|
|Environment:||Linux||Attachments:||( 0) Google-RTP.patch|
|Description:||Current code waits for RTP packet to come from google over RTP to setup the RTP handshake. This works most of the time except for the case of a dropped UDP packet. If the dropped packet is the initial udp packet sent from google on RTP then the call fails to establish properly. The workaround for this problem involves extracting the google RTP address from the candidate information presented from google on the JABBER channel and sending it to RTP. My patch seems to work but I am sure something more elegant could be crafted.|
|Comments:||By: Jonathan Rose (jrose) 2012-01-27 11:32:14.579-0600|
This isn't working at all for me. When I use this patch, all of my GTalk calls just lose all of their audio.
By: Malcolm Davenport (mdavenport) 2015-02-26 09:37:29.827-0600
Unfortunately, this issue wasn't addressed during the bug-fix lifetime of Asterisk 1.8. The good news is that Asterisk 11 and greater have chan_motif and res_xmpp, which are a rewrite of XMPP support within Asterisk, and are supported.
We'd encourage you to try that out instead and see if that clears things up for you.