[SR-Users] WebRTC to SIP gateway

[Yuriy Gorlichenko]

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 at 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 at DOMAIN>.
>
> From: <sip:4777519304 at DOMAIN>;tag=9qhqhrnj3s.
>
> Call-ID: j5h7830ivr5dfc2mn5sov1.
>
> CSeq: 4 REGISTER.
>
> Contact: <sip:c3qkm4fv at 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 at DOMAIN>;tag=91334f57.
>
> From: <sip:4777519304 at 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 at phonect.no
> Mobile: +47 41 88 88 00
>
>
>  [image: cid:image007.png at 01D3A0E8.376921D0] <http://www.phonect.no/>
>
> [image: facebook_2] <https://www.facebook.com/phonectno> [image:
> LinkedIn_logo_initials (1)]
> <https://www.linkedin.com/company/44983?trk=tyah&trkInfo=clickedVertical%3Acompany%2CentityType%3AentityHistoryName%2CclickedEntityId%3Acompany_company_44983%2Cidx%3A0>
>
>
>
> _______________________________________________
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