[Freeswitch-users] RTP problems in recent revisions?

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.