[SR-Users] WebRTC ACK Protocol

[Jorge Cornejo]

Can you share your config file?

 

From: sr-users [mailto:sr-users-bounces at lists.kamailio.org] On Behalf Of Ilie Soltanici
Sent: quarta-feira, 3 de abril de 2019 14:34
To: Kamailio (SER) - Users Mailing List <sr-users at lists.kamailio.org>
Subject: [SR-Users] WebRTC ACK Protocol

 

Hello,

 

I am trying to set up a WebRTC2SIP Gateway by using Kamailio and rtpengine. So far, everything is working fine, I'm able to register an extension and make a call, but for some reason, when i'm trying to call a WebRTC extension from any SIP Extension Kamailio is sending INVITE, WebRTC extension is sending back 200 OK, and then Kamailio is trying to send an ACK through UDP protocol, and not through wss, as it's supposed to do. This is how invite is looking:

 

INVITE sip:nl7oe4ss at vjbh7r4im6j7.invalid;transport=wss SIP/2.0

Record-Route: <sip:my-company.net <http://my-company.net> ;transport=udp;ftag=as1789445c;lr=on;nat=yes>

Via: SIP/2.0/WSS 123.123.123.123:10443;branch=z9hG4bKe655.29d7c135a302f3eb803902d4f5a8da7e.0

Via: SIP/2.0/UDP 192.168.50.237:5060;received=192.168.50.237;branch=z9hG4bK7d2e534e;rport=5060

Max-Forwards: 70

From: "WebRTC" <sip:11 at my-company.net <mailto:sip%3A11 at my-company.net> >;tag=as1789445c

To: <sip:15 at 192.168.50.210:5060 <http://sip:15@192.168.50.210:5060> >

Contact: <sip:11 at 192.168.50.237:5060 <http://sip:11@192.168.50.237:5060> >

Call-ID: 7fc800de060197fa2315c93763873092 at my-company.net <mailto:7fc800de060197fa2315c93763873092 at my-company.net> 

CSeq: 102 INVITE

User-Agent: Proxy

Date: Wed, 03 Apr 2019 17:11:41 GMT

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

Supported: replaces, timer

Alert-Info:

Content-Type: application/sdp

Content-Length: 596

Server: SIP Proxy

 

and then WebRTC app is replying with 200 OK:

 

SIP/2.0 200 OK

Record-Route: <sip:my-company.net <http://my-company.net> ;transport=udp;ftag=as1789445c;lr=on;nat=yes>

Via: SIP/2.0/WSS 123.123.123.123:10443;branch=z9hG4bKe655.29d7c135a302f3eb803902d4f5a8da7e.0

Via: SIP/2.0/UDP 192.168.50.237:5060;received=192.168.50.237;branch=z9hG4bK7d2e534e;rport=5060

To: <sip:15 at 192.168.50.210:5060 <http://sip:15@192.168.50.210:5060> >;tag=dk4fa8ftt6

From: "WebRTC" <sip:11 at my-company.net <mailto:sip%3A11 at my-company.net> >;tag=as1789445c

Call-ID: 7fc800de060197fa2315c93763873092 at my-company.net <mailto:7fc800de060197fa2315c93763873092 at my-company.net> 

CSeq: 102 INVITE

Contact: <sip:nl7oe4ss at vjbh7r4im6j7.invalid;transport=wss>

Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER

Supported: outbound

User-Agent: Proxy-WEBRTC

Content-Type: application/sdp

Content-Length: 901

 

and finally, Kamailio is trying to send this ack through UDP protocol:

 

ACK sip:nl7oe4ss at 22.22.22.22:57421;transport=wss SIP/2.0

Via: SIP/2.0/UDP 192.168.50.237:5060;branch=z9hG4bK56363ddf;rport

Route: <sip:my-company.net <http://my-company.net> ;transport=udp;ftag=as1789445c;lr=on;nat=yes>

Max-Forwards: 70

From: "WebRTC" <sip:11 at my-company.net <mailto:sip%3A11 at my-company.net> >;tag=as1789445c

To: <sip:15 at 192.168.50.210:5060 <http://sip:15@192.168.50.210:5060> >;tag=dk4fa8ftt6

Contact: <sip:11 at 192.168.50.237:5060 <http://sip:11@192.168.50.237:5060> >

Call-ID: 7fc800de060197fa2315c93763873092 at my-company.net <mailto:7fc800de060197fa2315c93763873092 at my-company.net> 

CSeq: 102 ACK

User-Agent: Proxy

Content-Length: 0

 

If i'm trying to force it through TLS, i'm receiving error: 

get_send_socket2(): protocol/port mismatch (forced tls:123.123.123.123:10443 <http://123.123.123.123:10443> , to udp:22.22.22.22:23317 <http://22.22.22.22:23317> )

 

Can someone point me in the right direction, please?

Thank you.

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20190403/d6f836d7/attachment.html>