Hello, I need implement one way call without RTP stream on one side. I need it for reduce bandwith. Turn off rtp stream is not necessary just audio data should be zero. I'm not expert in this area and i don't know how can i realize this feature. I'm try make "call on hold" which send new SDP message with audio atribute "sendonly" when connection is confirmed ...but not success. - audio is not heard but RTP stream is still sending. - PBX asterisk do not send "sendonly" message to callee because is a B2BUA, it sits in the middle and when end puts the call on hold it provides the on hold audio Situation is following: 1) A send invite message with new SDP "sendonly" A ---- sendonly --> PBX 2) PBX response to A with "recvonly" PBX ---recvonly ---> A 3) PBX send invite to B new SDP "sendrecv" PBX ---sendrecv ---> B 4) B response with sendrecv too B ---sendrecv ---> PBX here is log from asterisk: http://pastebin.org/254229 In RFC 3264 is: " If the offerer wishes to only receive media from its peer, it MUST mark the stream as recvonly. If the offerer wishes to communicate, but wishes to neither send nor receive media at this time, it MUST mark the stream with an "a=inactive" attribute. The inactive direction attribute is specified in RFC 3108 [3]. Note that in the case of the Real Time Transport Protocol (RTP) [4], RTCP is still sent and received for sendonly, recvonly, and inactive streams. " So set SDP attribute to sendonly, recvonly will be not work for me...media are still sending . Maybe when i will be check if "sendonly" attribute is receive and then i disable sending data to RTP stream. But PBX will not re- send my request to callee..maybe signaling in SIP protocol..exists some mechanism how can i realize this? thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20100519/2b5343eb/attachment.html>
