Hello!

It's been several years since I've used Kamailio. My current employer wants to 
implement WebRTC, which is currently not supported in our SIP backend, and 
asked if I could set up a Kamailio server as a gateway.

I've been able to make calls in all directions between SIP and WebRTC clients 
registered locally on Kamailio. When I tried to connect the server to the SIP 
backend, I ran into an issue. I'm able to register SIP clients in the backend 
via the gateway and make calls everywhere. However, the WebRTC client fails to 
register. Here are the messages between the Kamailio gateway and the SIP 
backend:


U 2018/05/09 10:12:58.316643 GATEWAY:15060 -> DOMAIN:5060
REGISTER sip:DOMAIN SIP/2.0.
Via: SIP/2.0/UDP 
GATEWAY:15060;branch=z9hG4bK4fc6.04d1730be5b78d595c69a3aa137987c1.0.
Via: SIP/2.0/AUTO 
lr2l9s72ehhc.invalid;rport=61353;received=CLIENT;branch=z9hG4bK5927151.
Max-Forwards: 68.
To: <sip:4777519304@DOMAIN>.
From: <sip:4777519304@DOMAIN>;tag=9qhqhrnj3s.
Call-ID: j5h7830ivr5dfc2mn5sov1.
CSeq: 4 REGISTER.
Contact: 
<sip:c3qkm4fv@lr2l9s72ehhc.invalid;transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:0ef2deac-2d56-465a-840b-543b9fd01af8>";expires=600.
Expires: 600.
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO.
Supported: path,gruu,outbound.
User-Agent: JsSIP 3.2.9.
Content-Length: 0.
Path: <sip:GATEWAY:15060;lr>.
.


U 2018/05/09 10:12:58.368409 DOMAIN:5060 -> GATEWAY:15060
SIP/2.0 400 Wrong transport. Provided transport either invalid or not 
supported..
Via: SIP/2.0/UDP 
GATEWAY:15060;branch=z9hG4bK4fc6.04d1730be5b78d595c69a3aa137987c1.0.
Via: SIP/2.0/AUTO 
lr2l9s72ehhc.invalid;rport=61353;received=CLIENT;branch=z9hG4bK5927151.
To: <sip:4777519304@DOMAIN>;tag=91334f57.
From: <sip:4777519304@DOMAIN>;tag=9qhqhrnj3s.
Call-ID: j5h7830ivr5dfc2mn5sov1.
CSeq: 4 REGISTER.
Content-Length: 0.


I believe that this error message is caused by ';transport=ws' in the Contact 
header. I'm not allowed to modify this header.

In the backend database, I found that some other clients have ';transport=UDP' 
in their path headers, so I tried to add that. (Why can I not add parameters in 
path module without adding username?) I still got the same error.

How do I best proceed?

For your information: We have outsourced the development of the WebRTC client, 
so we are able to change it. We also have the option of paying the supplier of 
the backend for development there.


With kind regards
Pan B. Christensen
Developer

Phonect AS
Brugata 19, PB 9156 Grønland, N-0133 Oslo, Norway
E-mail: pan.christen...@phonect.no<mailto:pan.christen...@phonect.no>
Mobile: +47 41 88 88 00


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