[SR-Users] Kamailio as Websocket bridge to Asterisk, and Asterisk-sent OPTIONS

[Paul Belanger]

On Thu, Aug 28, 2014 at 7:18 PM, Alex Villací­s Lasso
<a_villacis at palosanto.com> wrote:
> As a continuation of my project, I am trying to set up Kamailio as a
> Websocket bridge to Asterisk. The asterisk instance is running as localhost,
> with its own websocket support disabled, but otherwise has accounts with all
> of the avfp and dtls settings for websockets. Additionally, I have removed
> the bindaddr=127.0.0.1 from sip.conf and instead put a deny=0.0.0.0/0.0.0.0
> and permit=127.0.0.1/255.255.255.0 in order to restrict SIP signaling to
> localhost. This allows asterisk to bypass rtpproxy when signaling through a
> websocket. I have already established calls originating from the browser.
> However, I have an issue with the registration.
>
Just in passing, why did you remove bindaddr=127.0.0.1?

> In my setup, Kamailio receives the REGISTER from whatever source, and
> forwards this through UDP to Asterisk, after the multiple-domain
> transformation. Therefore, Asterisk sees the following in its SIP port (all
> traffic through localhost):
>
> REGISTER sip:pbx.villacis.com SIP/2.0
> Via: SIP/2.0/UDP
> 127.0.0.1;branch=z9hG4bKc1c5.cb49f656197d0ba16f2a1661dd6a44cc.0
> Via: SIP/2.0/WSS
> r01r0mla9hdp.invalid;rport=47307;received=192.168.3.2;branch=z9hG4bK9309681
> Max-Forwards: 69
> To: <sip:avillacisIM_pbx.villacis.com at 127.0.0.1:5080>
> From: "Alex Villac..s"
> <sip:avillacisIM_pbx.villacis.com at 127.0.0.1:5080>;tag=b5c0lq4kac
> Call-ID: vp2akar0aqfmgfa6m1taau
> CSeq: 82 REGISTER
> Contact:
> <sip:fnuql6ft at 192.168.3.2:47307;transport=ws>;reg-id=1;+sip.instance="<urn:uuid:6b0c58ee-bdc5-47c0-aff0-963132dc0cad>";expires=600
> Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY,INVITE,MESSAGE
> Supported: path,gruu,outbound
> User-Agent: SIP.js/0.6.2
> Content-Length: 0
>
> Asterisk answers this through UDP, and Kamailio forwards it through the
> websocket:
>
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 127.0.0.1;branch=z9hG4bKc1c5.cb49f656197d0ba16f2a1661dd6a44cc.0;received=127.0.0.1;rport=5060
> Via: SIP/2.0/WSS
> r01r0mla9hdp.invalid;rport=47307;received=192.168.3.2;branch=z9hG4bK9309681
> From: "Alex Villac..s"
> <sip:avillacisIM_pbx.villacis.com at 127.0.0.1:5080>;tag=b5c0lq4kac
> To: <sip:avillacisIM_pbx.villacis.com at 127.0.0.1:5080>;tag=as5ae2df76
> Call-ID: vp2akar0aqfmgfa6m1taau
> CSeq: 82 REGISTER
> Server: Asterisk PBX 11.12.0
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH, MESSAGE
> Supported: replaces, timer
> Expires: 600
> Contact: <sip:fnuql6ft at 192.168.3.2:47307;transport=ws>;expires=600
> Date: Thu, 28 Aug 2014 22:21:15 GMT
> Content-Length: 0
>
> Then Asterisk sends this through UDP, and Kamailio again forwards it through
> the websocket:
>
> NOTIFY sip:fnuql6ft at 192.168.3.2:47307;transport=ws SIP/2.0
> Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK4d60f167;rport
> Max-Forwards: 70
> From: "asterisk" <sip:asterisk at 127.0.0.1:5080>;tag=as43c12840
> To: <sip:fnuql6ft at 192.168.3.2:47307;transport=ws>
> Contact: <sip:asterisk at 127.0.0.1:5080>
> Call-ID: 04deeb0068a847fa514d748c7d9993c5 at 127.0.0.1:5080
> CSeq: 102 NOTIFY
> User-Agent: Asterisk PBX 11.12.0
> Event: message-summary
> Content-Type: application/simple-message-summary
> Content-Length: 89
>
> Messages-Waiting: no
> Message-Account: sip:*97 at 127.0.0.1:5080
> Voice-Message: 0/0 (0/0)
>
> Since I have not implemented handling of voicemail indications, the browser
> answers this:
>
> SIP/2.0 405 Method Not Allowed
> Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK4d60f167;rport=5080
> To: <sip:fnuql6ft at 192.168.3.2:47307;transport=ws>;tag=ggu5etber9
> From: "asterisk" <sip:asterisk at 127.0.0.1:5080>;tag=as43c12840
> Call-ID: 04deeb0068a847fa514d748c7d9993c5 at 127.0.0.1:5080
> CSeq: 102 NOTIFY
> Supported: outbound
> Content-Length: 0
>
>
> After that, Asterisk wants to send an OPTIONS packet. From the point of view
> of Asterisk (sip set debug on), it is already sent, but never gets a
> response. However, tcpdump shows that the packet is never sent through the
> localhost interface in the first place. It is also not sent through any
> other interface. My guess is that since the REGISTER has a contact with
> transport=ws , Asterisk wants to send this through a websocket (which is
> disabled). So I could have to generate a contact without transport=ws .
>
> I have worked around this by setting qualify=no in the account for the
> websocket, but I would like a better solution, one that allows the OPTIONS
> packet to reach the browser, and to get the response. What is the proper way
> to deal with this?
>
What does the OPTIONS message in asterisk look like?

-- 
Paul Belanger | PolyBeacon, Inc.
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