[Freeswitch-users] Issue with SDP - Call hangs up right after SIP 200 OK

[Marc Bernard]

Removing video codecs from SIP profile did solve the issue.

 

Thanks Valli.

 

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Just for reference, here is the offer to that Panasonic phone before the fix: 

 

)Hhe.E at C¸ª%æÙÄÄs3ºINVITE <mailto:Hhe.E at C¸ª%25æÙÄÄs3ºINVITE>  sip:Account1 at 1.1.217.127:5060;received=1.1.217.127:5060 SIP/2.0

Via: SIP/2.0/UDP 1.1.131.37;rport;branch=z9hG4bKttaHKmXNgBtSQ

Max-Forwards: 67

From: "Debug Proxy" <sip:+15145552353 at 1.1.131.37>;tag=rDQF79t14QmKr

To: <sip:9919605 at dev-proxy.xxx.com>

Call-ID: cbecb6ef-4300-1239-dfa0-000c29124868

CSeq: 22952030 INVITE

Contact: <sip:mod_sofia at 1.1.131.37:5060>

User-Agent: Acme CloudPBX

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE

Supported: timer, path, replaces

Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer

Session-Expires: 120;refresher=uac

Min-SE: 120

Content-Type: application/sdp

Content-Disposition: session

Content-Length: 397

X-DNIS: 9919605

Remote-Party-ID: "Debug Proxy" <sip:+15145552353 at 1.1.131.37>;party=calling;screen=yes;privacy=off

 

v=0

o=FreeSWITCH 1594981106 1594981107 IN IP4 1.1.131.37

s=FreeSWITCH

c=IN IP4 1.1.131.37

t=0 0

m=audio 30538 RTP/AVP 0 101 13

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtpmap:13 CN/8000

a=ptime:20

m=video 25520 RTP/AVP 34

b=AS:1024

a=rtpmap:34 H263/90000

a=rtcp-fb:34 ccm fir

a=rtcp-fb:34 ccm tmmbr

a=rtcp-fb:34 nack

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