pjsip offers 2 audio streams on invite with empty SDP

Alexander Butenko <a.butenka@xxxxxxxxx> · Mon, 20 Nov 2017 10:40:03 -0400

Hello guys,

Im using pjsip 2.5.5 and somehow on 
INVITE without an SDP im sending 200 OK with 2 SDP streams somehow. In 
the end im getting a call without any audio. Any ideas how I can avoid 
that? 

Thank you,
Alex

=== INVITE ===
INVITE sip:2379999@172.16.1.80 SIP/2.0
Via: SIP/2.0/UDP 172.16.1.80:5060;branch=z9hG4bKac1412631967
Max-Forwards: 10
From: <sip:XXXX@172.16.1.86>;tag=1c1412511978
To: <sip:2379999@172.16.1.86>
Call-ID: 17141348@172.16.1.80
CSeq: 1 INVITE
Contact: <sip:XXXXX@172.16.1.80:5060>
Supported: timer
Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
Session-Expires: 1800;refresher=uac
Min-SE: 90
User-Agent: Mediant 4000/v.6.80A.231.002
Content-Length: 0
X-Genesys-CallUUID: XXXXXX

== 200 OK from PJSIP ==
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.214.20.50;received=10.214.20.50;branch=z9hG4bK8c3c.2ef4e1a3132abdbe242000990263
8c.0
Via: SIP/2.0/UDP 172.16.1.80:5060;rport=5060;branch=z9hG4bKac1412631967
Record-Route: <sip:10.214.20.50;lr;nat=yes>
Call-ID: 141249285920112017141348@172.16.1.80
From: <sip:123@172.16.106.86>;tag=1c1412511978
To: <sip:2379999@172.16.106.86>;tag=f3e2708cdf5d4b48952fca438d7a0bd1
CSeq: 1 INVITE
Contact: <sip:2379999@10.214.70.31:52853;ob>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:   857

v=0
o=- 3720157800 3720157800 IN IP4 10.214.70.31
s=pjmedia
b=AS:151
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 98 97 99 104 3 0 8 9 96
c=IN IP4 10.214.70.31
b=TIAS:64000
a=rtcp:4001 IN IP4 10.214.70.31
a=sendrecv
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=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
m=audio 4002 RTP/AVP 98 97 99 104 3 0 8 9 96
c=IN IP4 10.214.70.31
b=TIAS:64000
a=rtcp:4003 IN IP4 10.214.70.31
a=sendrecv
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

-- 
Alex

_______________________________________________
Visit our blog: http://blog.pjsip.org

pjsip mailing list
pjsip@xxxxxxxxxxxxxxx
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org