[Freeswitch-users] RTP problems in recent revisions?

[Michael Jerris]

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 <jason at jasonjgw.net> 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.
>
>
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