[Freeswitch-users] RTP problems in recent revisions?
mike at jerris.com
Sat Dec 19 06:25:22 PST 2009
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.
On Dec 19, 2009, at 3:31 AM, Jason White <jason at jasonjgw.net> wrote:
> Revision 15904 is fine, but after upgrading to revision 16003 I get
> 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
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> FreeSWITCH-users at lists.freeswitch.org
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