9 May
2018
9 May
'18
4:09 p.m.
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<mailto:pan.christensen@phonect.no>
Mobile: +47 41 88 88 00
[cid:image007.png@01D3A0E8.376921D0] <http://www.phonect.no/>
[facebook_2]<https://www.facebook.com/phonectno> [LinkedIn_logo_initials (1)]
<https://www.linkedin.com/company/44983?trk=tyah&trkInfo=clickedVertical%3Acompany%2CentityType%3AentityHistoryName%2CclickedEntityId%3Acompany_company_44983%2Cidx%3A0>
9 May
9 May
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/>
[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>
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
11 May
11 May
4:28 p.m.
Hello again!
I've figured out that our SIP backend doesn't like headers with
';transport=ws' or Via: SIP/2.0/AUTO
I need to modify or remove contact, Via and Record-route headers.
I found that the TOPOS module is able to do this. I added it with default settings and
applied it to all messages going to the backend. I was then able to make an outgoing call,
but found that topology was only hidden in initial requests (not ACK, UPDATE, BYE etc.).
Applying the module to messages in all directions worked better.
Being able to make outgoing calls is good progress, but this client will mostly be used
for incoming calls (switchboard).
TOPOS does not modify REGISTER. I'm not sure if it's able to or if it should.
We tried modifying the headers in the client, but then Kamailio replies '478
Unresolvable destination' to INVITE from the backend.
How can I enable the WebRTC client to register against the backend and maintain correct
routing? Should TOPOS be able to do this? Can I use another module? Is it possible to
register the client in Kamailio and then register Kamailio with the backend (building a
new REGISTER message)? Any other suggestions?
With kind regards
Pan B. Christensen
Developer
Phonect AS
From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Pan Christensen
Sent: onsdag 9. mai 2018 15:09
To: sr-users(a)lists.kamailio.org
Subject: [SR-Users] WebRTC to SIP gateway
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<mailto:pan.christensen@phonect.no>
Mobile: +47 41 88 88 00
[cid:image007.png@01D3A0E8.376921D0] <http://www.phonect.no/>
[facebook_2]<https://www.facebook.com/phonectno> [LinkedIn_logo_initials (1)]
<https://www.linkedin.com/company/44983?trk=tyah&trkInfo=clickedVertical%3Acompany%2CentityType%3AentityHistoryName%2CclickedEntityId%3Acompany_company_44983%2Cidx%3A0>
2387
days inactive
2389
days old
2 comments
2 participants
participants (2)
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Pan Christensen -
Yuriy Gorlichenko