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@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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