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

[Alex Villací­s Lasso]

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.

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?