Hello, On 9/3/07, li zhang <[EMAIL PROTECTED]> wrote: > when i use sipXezphone and placecall demo built on sipXtapi 2.9.1.0 from svn > branch,sometimes can not send and receive rtp stream. when i capture packet > by wireshark, i find sipphone send a rtp packet which version is 00 and > proxy(asterisk) return a rtp packet which version is 00 too. After that rtp > stream is stopped.
Packets with RTP version set to 0 are STUN/TURN packets. May be Asterisk report you wrong STUN/TURN data or it is incorrectly handled by sipXtapi. Wireshark should be able to decode STUN packets. I recall it have option how to decode packets with RTP version 0, and one of choices is a STUN packet. -- Regards, Alexander Chemeris. SIPez LLC. SIP VoIP, IM and Presence Consulting http://www.SIPez.com tel: +1 (617) 273-4000 _______________________________________________ sipxtapi-dev mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipxtapi-dev/
