[Freeswitch-users] RTP problems in recent revisions?

[Jason White]

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.