Hi! 
You can to hande it with add_contact_alias but im not sure it will rewrite transport for you

also if you will store contact as it is on your backend it is a big chance that it can be unusefull with your SIP service because conract uri is encrypted and most of b2b servers like asterisk for example will can not resolve this.

Only one I can suggest - is rewrite contact header with ip address of webrtc2sip gateway you are building

May be some one else will can suggest some more usefull solutions for you.

On Wed, May 9, 2018, 16:11 Pan Christensen <pan.christensen@phonect.no> wrote:

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.christensen@phonect.no
Mobile: +47 41 88 88 00


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