Re: [Freeswitch-users] RTP problems in recent revisions?
I tried a patch out of pure deduction and speculation from your post. Can you update and test it for me please? On Sat, Dec 19, 2009 at 9:19 AM, Anthony Minessale < anthony.miness...@gmail.com> wrote: > Also retest with no zrtp > send a full console debug log with sip trace > > On Dec 19, 2009 8:33 AM, "Michael Jerris" wrote: > > The best help to track this down is to try to identify the specific > svn revision that caused the issue and to supply a full freeswitch > debug with sip trace. > > Mike > > On Dec 19, 2009, at 3:31 AM, Jason White wrote: > > Revision 15904 is fine, but... > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org pstn:+19193869900 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] RTP problems in recent revisions?
Also retest with no zrtp send a full console debug log with sip trace On Dec 19, 2009 8:33 AM, "Michael Jerris" wrote: The best help to track this down is to try to identify the specific svn revision that caused the issue and to supply a full freeswitch debug with sip trace. Mike On Dec 19, 2009, at 3:31 AM, Jason White wrote: > Revision 15904 is fine, but... ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] RTP problems in recent revisions?
The best help to track this down is to try to identify the specific svn revision that caused the issue and to supply a full freeswitch debug with sip trace. Mike On Dec 19, 2009, at 3:31 AM, Jason White wrote: > Revision 15904 is fine, but after upgrading to revision 16003 I get > the > following. > > 1. No problems with a FreeSWITCH to FreeSWITCH call (Celt codec). > > 2. A PCMU call to a SIP provider is fine for the first 20 to 30 > seconds, then > the audio breaks up completely. > > I have ZRTP compiled in, if that makes any difference. > > Obviously there's a regression somewhere. Let me know if I can > provide further > help. > > > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] RTP problems in recent revisions?
Revision 15904 is fine, but after upgrading to revision 16003 I get the following. 1. No problems with a FreeSWITCH to FreeSWITCH call (Celt codec). 2. A PCMU call to a SIP provider is fine for the first 20 to 30 seconds, then the audio breaks up completely. I have ZRTP compiled in, if that makes any difference. Obviously there's a regression somewhere. Let me know if I can provide further help. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
