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

[Alex Villací­s Lasso]

El 28/08/14 19:09, Paul Belanger escribió:
> 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?
If I keep the bindaddr, then asterisk fails to send the DTLS-SRTP handshake packets, resulting in no audio. Apparently rtpproxy does not route this.
>> 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?
>

elx3*CLI> sip qualify peer avillacisIM_pbx.villacis.com
Reliably Transmitting (NAT) to 127.0.0.1:5060:
OPTIONS sip:68on862t at 192.168.3.2:58927;transport=ws SIP/2.0
Via: SIP/2.0/WS 127.0.0.1:5080;branch=z9hG4bK2b267794;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 127.0.0.1:5080>;tag=as1a2c3be2
To: <sip:68on862t at 192.168.3.2:58927;transport=ws>
Contact: <sip:asterisk at 127.0.0.1:5080;transport=WS>
Call-ID: 7cbd63985b293b0150740e5a19143451 at 127.0.0.1:5080
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.12.0
Date: Fri, 29 Aug 2014 15:54:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


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