thanks your reply! but i not enable STUN/TURN service. Best wishes, Li Zhang
2007/9/3, Alexander Chemeris <[EMAIL PROTECTED]>: > > 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 >
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