9 May
2018
9 May
'18
4:52 p.m.
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(a)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(a)phonect.no
Mobile: +47 41 88 88 00
[image: cid:image007.png@01D3A0E8.376921D0] <http://www.phonect.no/>
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