Re: [SIP.js] Re: Wrong Contact header on SIP 200 OK

[Guna Ramanujam]

Gist:

https://gist.github.com/57255016fe4deec15665.git

Includes log from register and multiple calls. Thought you might need all, incase. To jump right to the log point where issue occurs, search for “Request-URI does not point to us”

Guna

From: Eric Green
Sent: ‎Wednesday‎, ‎March‎ ‎4‎, ‎2015 ‎2‎:‎14‎ ‎AM
To: sip...@googlegroups.com

Hi Guna,

Can you please attach a gist with full javascript logs and traceSip turned on?

Thanks.

-Eric Green

On Friday, February 27, 2015 at 12:13:03 PM UTC-5, Guna Ramanujam wrote:

Bob's leg of the call seems to be generating wrong and multiple Contact header in response to INVITE. This only happens spuriously on Chrome. This is causing response ACK from peer to be ignored with: <b>Request-URI does not point to us</b>

Any idea why this may be?

Log:

INVITE sip:7e9e...@192.0.2.28;transport=ws SIP/2.0
Via: SIP/2.0/WS XX.XX.XX.XX:0;branch=z9hG4bK65839aee;rport
Max-Forwards: 70
From: "1083" <sip:10...@XX.XX.XX.XX:0>;tag=as310863e4
To: <sip:7e9e...@192.0.2.28;transport=ws>
Contact: <sip:10...@XX.XX.XX.XX:0;transport=WS>
Call-ID: 1f9a1f4c096f282343075dba5c38b72...@XX.XX.XX.XX:0
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.2.0
Date: Fri, 27 Feb 2015 11:47:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 1086
.....................

SIP/2.0 100 Trying
Via: SIP/2.0/WS XX.XX.XX.XX:0;branch=z9hG4bK65839aee;rport
To: <sip:7e9e...@192.0.2.28;transport=ws>
From: "1083" <sip:10...@XX.XX.XX.XX:0>;tag=as310863e4
Call-ID: 1f9a1f4c096f282343075dba5c38b72...@XX.XX.XX.XX:0
CSeq: 102 INVITE
Supported: outbound
Content-Length: 0

SIP/2.0 180 Ringing
Via: SIP/2.0/WS XX.XX.XX.XX:0;branch=z9hG4bK65839aee;rport
To: <sip:7e9e...@192.0.2.28;transport=ws>;tag=iqi4u1lvlh
From: "1083" <sip:10...@XX.XX.XX.XX:0>;tag=as310863e4
Call-ID: 1f9a1f4c096f282343075dba5c38b72...@XX.XX.XX.XX:0
CSeq: 102 INVITE
Contact: <sip:7e9e...@192.0.2.28;transport=ws>
Supported: outbound
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/WS XX.XX.XX.XX:0;branch=z9hG4bK65839aee;rport
To: <sip:7e9e...@192.0.2.28;transport=ws>;tag=iqi4u1lvlh
From: "1083" <sip:10...@XX.XX.XX.XX:0>;tag=as310863e4
Call-ID: 1f9a1f4c096f282343075dba5c38b72...@XX.XX.XX.XX:0
CSeq: 102 INVITE
Contact: <sip:l1po...@192.0.2.52;transport=ws;ob>
Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY,INVITE,MESSAGE
Content-Type: application/sdp
Contact: <sip:l1po...@192.0.2.52;transport=ws;ob>
Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY,INVITE,MESSAGE
Content-Type: application/sdp
Contact: <sip:7e9e...@192.0.2.28;transport=ws>
Supported: outbound
Content-Type: application/sdp
Content-Length: 1854
.............

ACK sip:l1po...@192.0.2.52;transport=ws;ob SIP/2.0
Via: SIP/2.0/WS XX.XX.XX.XX:0;branch=z9hG4bK5d19cbe8;rport
Max-Forwards: 70
From: "1083" <sip:10...@XX.XX.XX.XX:0>;tag=as310863e4
To: <sip:7e9e...@192.0.2.28;transport=ws>;tag=iqi4u1lvlh
Contact: <sip:10...@XX.XX.XX.XX:0;transport=WS>
Call-ID: 1f9a1f4c096f282343075dba5c38b72...@XX.XX.XX.XX:0
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.2.0
Content-Length: 0

sip-0.6.4.js:2680 Fri Feb 27 2015 17:16:54 | sip.ua | Request-URI does not point to us

-g

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[Eric Green]

I looked at the logs. I am not an Asterisk expert by any means, so maybe some other people here could shed some more light on this. It looks like Asterisk is not respecting the contact that you send it. By mangling this it will definitely throw an error and rightfully show. Asterisk should be respecting the contact headers. I am not sure why Asterisk is doing this. Following up with Asterisk might be necessary.

-Eric Green

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