[OpenSIPS-Users] MediaProxy do not add Relay Candidate

vivid333 vivid333 at 163.com
Thu Dec 1 08:09:03 CET 2011


Hello:

             A->opensips (+mediaproxy)-> B

             Can any one help me why MediaProxy don't add Relay 
candidate? I am sure that use_media_proxy works before  opensips send 
invite to B.


////////////////////////////A(pjsip) Send Invite,  B side receive Invite 
include relay candidate
*********************************************************************************************
INVITE sip:TEST at 225.4.xxx.xxx:5060 SIP/2.0
Via: SIP/2.0/UDP 
114.249.xxx.xxx:32768;rport;branch=z9hG4bKPjRJalhnJGIbBwiSvzyqUo2rgOM0ld3Wqo
Max-Forwards: 70
From: sip:+8618601036573 at 225.4.xxx.xxx;tag=qv4WrTe2HkdTK8U2W8Y40SgAeHIehYGR
To: sip:TEST at 225.4.xxx.xxx
Contact: <sip:+8618601036573 at 114.246.xxx.xxx:32768;ob>
Call-ID: xxAb8TPG-d-ePhIOx1Xw6jyDjywCJDeU
CSeq: 23079 INVITE
Route: <sip:225.4.xxx.xxx:5060;lr>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, 
REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: PJSUA v1.10.0 Linux-2.6.32.35/i686/glibc-2.11
Content-Type: application/sdp
Content-Length:   629

v=0
o=- 3531703180 3531703180 IN IP4 114.246.xxx.xxx
s=pjmedia
c=IN IP4 114.246.xxx.xxx
t=0 0
a=X-nat:7
m=audio 56846 RTP/AVP 98 97 99 104 3 0 8 9 96
a=rtpmap:98 speex/16000
a=rtpmap:97 speex/8000
a=rtpmap:99 speex/32000
a=rtpmap:104 iLBC/8000
a=fmtp:104 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=sendrecv
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=ice-ufrag:025fe5d4
a=ice-pwd:72314495
a=candidate:Sa00020f 1 UDP 1862270975 114.246.xxx.xxx 56846 typ srflx 
raddr 10.0.2.15 rport 49367
a=candidate:Ha00020f 1 UDP 1694498815 10.0.2.15 49367 typ host


////////////////////////////A(linphone). change Pjsip to linphone,  B 
side receive Invite have no relay candidate
*********************************************************************************************
INVITE sip:TEST at 225.4.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 10.0.2.15:41338;rport;branch=z9hG4bK2713821050
From: <sip:vivid333 at 225.4.xxx.xxx>;tag=2642865755
To: "TEST" <sip:TEST at 225.4.xxx.xxx>
Call-ID: 1915536026
CSeq: 20 INVITE
Contact: <sip:vivid333 at 10.0.2.15:41338>
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, 
SUBSCRIBE, INFO
Max-Forwards: 70
User-Agent: Linphone
Supported: replaces, 100rel, timer, norefersub
Content-Length:   413

v=0
o=- 3531651689 3531651689 IN IP4 114.246.xxx.xxx
s=linphone
c=IN IP4 114.246.xxx.xxx
t=0 0
a=ice-ufrag:18e62b54
a=ice-pwd:5cf02fda
a=candidate:Sa00020f 1 UDP 1862270975 114.246.xxx.xxx 46909 typ srflx 
raddr 10.0.2.15 rport 26836
a=candidate:Ha00020f 1 UDP 1694498815 10.0.2.15 26836 typ host
m=audio 46909 RTP/AVP 0 96
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=sendrecv








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