Registration fails with 403 Forbidden with linphone

[Andrew]

Hello, I have been testing sipjs with some 3rd party sip services, such as linphone, and find myself running into the following registration issue. I do not have the issue using X-Lite.

I used http://sipjs.com/demo-phone/

I then modified sipjs.6.3 and tried different combinations of the following:

- Setting hackViaTcp: true and hackIpInContact: true

- Changing the Via, Contact, Allow to match that of what X-Lite used, although I faked the ip and port in those tests.

I am using ws://edge.sip.onsip.com for wsServers

Might this have something to do with the onsip websocket relay/translation? I dont know how my websocket register is turned into a request for the linphone server. If the 403 was sent by the liphone server then it obviously received the to/from user name correctly but I have no way to know why it is rejecting the request. One thought is because of a blacklisted ip or port.

I verified the sip registration and response using tcpdump and it matches the sipjs trace pasted below.

Any ideas?

....... Using Sip.js(0.6.3) ........

REGISTER sip:sip.linphone.org SIP/2.0

Via: SIP/2.0/TCP 192.0.2.125;branch=z9hG4bK7653063

Max-Forwards: 70

To: "sipjstest1"<sip:sipjs...@sip.linphone.org>

From: "sipjstest1"<sip:sipjs...@sip.linphone.org>;tag=9sk5k95jj6

Call-ID: fkddq2anabtd8roc104hdv

CSeq: 81 REGISTER

Contact: <sip:g0j5...@192.0.2.125;transport=ws>;reg-id=1;+sip.instance="<urn:uuid:3a173fca-9b97-47ff-8e36-431c710ae5>"

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO

Expires: 3600

Supported: path,gruu,100rel,outbound

User-Agent: SIP.js/0.6.3

Content-Length: 0

Wed Dec 03 2014 19:24:38 GMT+0000 (UTC) | sip.ua | emitting event connected

Wed Dec 03 2014 19:24:38 GMT+0000 (UTC) | sip.transport | received WebSocket text message:

SIP/2.0 403 Forbidden

Via: SIP/2.0/TCP 192.0.2.125;branch=z9hG4bK7653063

To: "sipjstest1"<sip:sipjs...@sip.linphone.org>;tag=1ee758eef3cbb66f4da79e04a0e4dad7.9a05

From: "sipjstest1"<sip:sipjs...@sip.linphone.org>;tag=9sk5k95jj6

Call-ID: fkddq2anabtd8roc104hdv

CSeq: 81 REGISTER

Server: OpenSIPS (1.10.0-notls (x86_64/linux))

Content-Length: 0

...... Using X-Lite, 401 received as expected with digest ......

Session Initiation Protocol (401)

Via: SIP/2.0/UDP 192.168.1.4:16546;branch=z9hG4bK-d8754z-c50d0c3e3bf10f73-1---d8754z-;rport

Max-Forwards: 70

Contact: <sip:sipjs...@192.168.1.4:16546;rinstance=23381940ec1f7a53>

To: "sipjstest1"<sip:sipjs...@sip.linphone.org>

From: "sipjstest1"<sip:sipjs...@sip.linphone.org>;tag=19070d13

Call-ID: MDA3OTQ0NDFhZjliMmNjYTQ4MmFmNTA5YTlkMjEwNTg

CSeq: 1 REGISTER

Expires: 3600

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO

User-Agent: X-Lite 4.7.1 74247-3759ac27-W6.1

Content-Length: 0

Via: SIP/2.0/UDP 192.168.1.4:16546;branch=z9hG4bK-d8754z-c50d0c3e3bf10f73-1---d8754z-;rport=16546

From: "sipjstest1" <sip:sipjs...@sip.linphone.org>;tag=19070d13

To: "sipjstest1" <sip:sipjs...@sip.linphone.org>;tag=eXHDD93ZD5r2N

Call-ID: MDA3OTQ0NDFhZjliMmNjYTQ4MmFmNTA5YTlkMjEwNTg

CSeq: 1 REGISTER

Server: Flexisip/0.8.0 (sofia-sip-nta/2.0)

WWW-Authenticate: Digest realm="sip.linphone.org", nonce="meMp2AAAAABE3DEjAABdwFmC5mMAAAAA", opaque="+GNywA==", algorithm=MD5, qop="auth"

Content-Length: 0

Thanks,

Andrew

[Will Mitchell]

Hi Andrew,

You can only use the OnSIP WebSocket server to register with OnSIP.  To send register requests to other third party services, you will need to connect either directly to a websocket server that service provides, or run your own which can proxy the request over UDP or TCP.  It is expected behavior that OnSIP will respond with a 403 Forbidden for REGISTER requests on domains it does not manage registrations for.

Thanks,

Will

On Wednesday, December 3, 2014 2:59:39 PM UTC-5, Andrew wrote:

Hello, I have been testing sipjs with some 3rd party sip services, such as linphone, and find myself running into the following registration issue. I do not have the issue using X-Lite.

I used http://sipjs.com/demo-phone/

I then modified sipjs.6.3 and tried different combinations of the following:

- Setting hackViaTcp: true and hackIpInContact: true

- Changing the Via, Contact, Allow to match that of what X-Lite used, although I faked the ip and port in those tests.

I am using ws://edge.sip.onsip.com for wsServers

Might this have something to do with the onsip websocket relay/translation? I dont know how my websocket register is turned into a request for the linphone server. If the 403 was sent by the liphone server then it obviously received the to/from user name correctly but I have no way to know why it is rejecting the request. One thought is because of a blacklisted ip or port.

I verified the sip registration and response using tcpdump and it matches the sipjs trace pasted below.

Any ideas?

....... Using Sip.js(0.6.3) ........

REGISTER sip:sip.linphone.org SIP/2.0

Via: SIP/2.0/TCP 192.0.2.125;branch=z9hG4bK7653063

Max-Forwards: 70

To: "sipjstest1"<sip:sipjstest1@sip.linphone.org>

From: "sipjstest1"<sip:sipjstest1@sip.linphone.org>;tag=9sk5k95jj6

Call-ID: fkddq2anabtd8roc104hdv

CSeq: 81 REGISTER

Contact: <sip:g0j5...@192.0.2.125;transport=ws>;reg-id=1;+sip.instance="<urn:uuid:3a173fca-9b97-47ff-8e36-431c710ae5>"

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO

Expires: 3600

Supported: path,gruu,100rel,outbound

User-Agent: SIP.js/0.6.3

Content-Length: 0

Wed Dec 03 2014 19:24:38 GMT+0000 (UTC) | sip.ua | emitting event connected

Wed Dec 03 2014 19:24:38 GMT+0000 (UTC) | sip.transport | received WebSocket text message:

SIP/2.0 403 Forbidden

Via: SIP/2.0/TCP 192.0.2.125;branch=z9hG4bK7653063

To: "sipjstest1"<sip:sipjstest1@sip.linphone.org>;tag=1ee758eef3cbb66f4da79e04a0e4dad7.9a05

From: "sipjstest1"<sip:sipjstest1@sip.linphone.org>;tag=9sk5k95jj6

Call-ID: fkddq2anabtd8roc104hdv

CSeq: 81 REGISTER

Server: OpenSIPS (1.10.0-notls (x86_64/linux))

Content-Length: 0

...... Using X-Lite, 401 received as expected with digest ......

Session Initiation Protocol (401)

Via: SIP/2.0/UDP 192.168.1.4:16546;branch=z9hG4bK-d8754z-c50d0c3e3bf10f73-1---d8754z-;rport

Max-Forwards: 70

Contact: <sip:sipjs...@192.168.1.4:16546;rinstance=23381940ec1f7a53>

To: "sipjstest1"<sip:sipjstest1@sip.linphone.org>

From: "sipjstest1"<sip:sipjstest1@sip.linphone.org>;tag=19070d13

[Andrew Pappas]

Well that certainly explains it :) Any recommendation on a proxy?

Thanks,

Andrew

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[Will Mitchell]

We use OverSIP internally.  Kamailio also supports WebSocket connections, although I have not tried it myself.  A softswitch like Asterisk or FreeSWITCH could be used, although the proxy options are likely going to be easier to set up.

-Will

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[Andrew Pappas]

Thanks again. I also just came across webrtc2sip which looks promising.

https://code.google.com/p/webrtc2sip/#SIP_Proxy

Andrew

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[Andrew Pappas]

Registering fine now using overSIP. The default configuration worked.

Andrew