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.
a=sendonly is, as you suspect, a unidirectional stream. The server says that it will send data and will not receive data, so the client must open a listening port. You're doing the right thing.
If you're not getting audio, it might be time to haul out an analysis tool like Wireshark to see if the server's actually sending any RTP data.