[Freeswitch-users] RTP problems in recent revisions?
jason at jasonjgw.net
Sat Dec 19 00:31:42 PST 2009
Revision 15904 is fine, but after upgrading to revision 16003 I get the
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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