[Freeswitch-users] Sometimes missing a few seconds of audio when CN is offered in INVITE (and solution)

Peter Olsson peter.olsson at visionutveckling.se
Wed Sep 26 23:56:07 MSD 2012


Thanks Tony, I will try it out tomorrow - I'll get back with the results!

/Peter
________________________________
Från: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Anthony Minessale [anthony.minessale at gmail.com]
Skickat: den 26 september 2012 21:28
Till: FreeSWITCH Users Help
Ämne: Re: [Freeswitch-users] Sometimes missing a few seconds of audio when CN is offered in INVITE (and solution)

Looking at the code, I can see a solution assuming your theory is correct.
I pushed a patch to HEAD if you care to test it.


On Wed, Sep 26, 2012 at 3:26 AM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
Hello everyone! I experience a strange issue for bridged calls from Lync -> FreeSWITCH -> Asterisk, it seems to be related to CN packets. I’m not sure if this a bug or not so I ask the question here first.

The problem is that the audio back from the Asterisk server is not bridged back to the originating client for the first few seconds (how long is different on different calls), instead FS seems to send CN-packets back. I’ve looked into a wireshark dump, and I see the audio is coming from the Asterisk server (using the correct IP and port), but FS doesn’t write the same packet to the other call leg (back to Lync), instead it sends a CN packet every second or so. My guess is that the timestamp is being handled wrong somehow, and FS believes that the real packet is too old (since it has already sent a CN packet with a higher timestamp?), and should not be written to the other leg, then after a while it sends a new CN packet and so on.

The solution is to set  “suppress_cng=true” before bridging the call to the Asterisk server, when this is done, the audio is always bridged correctly, and nothing is missing.

I have pcaps of both working calls (with the variable set) and nonworking calls, so if you believe it might be something that FS should handle differently I’ll submit this to Jira.

/Peter

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com<mailto:MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<mailto:PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org<mailto:sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
!DSPAM:5063553532761904613329!



Join us at ClueCon 2011 Aug 9-11, 2011
More information about the FreeSWITCH-users mailing list