On Fri, Aug 29, 2014 at 11:55 AM, Alex Villacís Lasso <a_villa...@palosanto.com> wrote: > El 28/08/14 19:09, Paul Belanger escribió: > >> On Thu, Aug 28, 2014 at 7:18 PM, Alex Villacís Lasso >> <a_villa...@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. > FWIW: I added a new setting into chan_sip, rptbindaddr[1], which allows you to no control the interface RTP binds too. Not sure if that helps in your setup or not.
>>> 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@127.0.0.1:5080> >>> From: "Alex Villac..s" >>> <sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080>;tag=b5c0lq4kac >>> Call-ID: vp2akar0aqfmgfa6m1taau >>> CSeq: 82 REGISTER >>> Contact: >>> >>> <sip:fnuql6ft@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@127.0.0.1:5080>;tag=b5c0lq4kac >>> To: <sip:avillacisIM_pbx.villacis.com@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@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@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@127.0.0.1:5080>;tag=as43c12840 >>> To: <sip:fnuql6ft@192.168.3.2:47307;transport=ws> >>> Contact: <sip:asterisk@127.0.0.1:5080> >>> Call-ID: 04deeb0068a847fa514d748c7d9993c5@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@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@192.168.3.2:47307;transport=ws>;tag=ggu5etber9 >>> From: "asterisk" <sip:asterisk@127.0.0.1:5080>;tag=as43c12840 >>> Call-ID: 04deeb0068a847fa514d748c7d9993c5@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@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@127.0.0.1:5080>;tag=as1a2c3be2 > To: <sip:68on862t@192.168.3.2:58927;transport=ws> > Contact: <sip:asterisk@127.0.0.1:5080;transport=WS> > Call-ID: 7cbd63985b293b0150740e5a19143451@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 > Ya, your via address is over the WS. What does your peer settings look like for avillacisIM_pbx.villacis.com ? [1] http://svnview.digium.com/svn/asterisk?view=revision&revision=422241 -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
