I am a newbie to sip/sdp world.
From my understanding of SDP protocol, if we define a=sendonly from sip server to client softphone, the softphone should open one RTP session for listening, but it should not send any RTP packets to destination. Am I correct?
In my case, I can not hear any sounds coming in, and there is a RTP stream to upload audio. Note: I am using the multicast address.
here is a SIP/SDP dump (from server to client softphone):
SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.8.219:5060;branch=z9hG4bK-d8754z-b394381274917501-1---d8754z-;rport=5060 From: ;tag=d67855ee To: ;tag=KQQHgQ93Sjg1F Call-ID: YTExMzkwZDdhMGM1NTJmMDJlMGFiYjgxMGI1ZDNmMDI. CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120623T054003Z~65b2f2d2e7+unclean~20120623T083401Z Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 265 v=0 o=FreeSWITCH 1340907341 1340907343 IN IP4 224.168.168.168 s=FreeSWITCH c=IN IP4 224.168.168.168 t=0 0 a=sendonly m=audio 34567 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20
I use another softphone to multicast sound(verify by wireshark) on that address and port. why can not I hear the sound?
by the way, softphone i am using xlite, the server is freeswitch.