The problem has been resolved after i readed the "not seeing outbound rtp packet" mail, thank you again. By the way, the packet with rtp version set to 0 is sure stun keepalive packet. Why do the client which not enable stun service send this keepalive packet?
Best wishes Li Zhang 2007/9/4, li zhang <[EMAIL PROTECTED]>: > > 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 2.9.1.0from > > > 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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