On Fri, Aug 29, 2014 at 5:08 PM, Alex Villacís Lasso <a_villa...@palosanto.com> wrote: > El 29/08/14 14:44, Paul Belanger escribió: > >> 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 >> > mysql> select * from sip where name = 'avillacisIM_pbx.villacis.com'; > +----+------------------------------+--------------------------------+-------------+-----------------+-----------------+------+--------+-----------+--------------+------------+---------+---------------------+--------+-------------+----------+-----------+-------------+----------------+------------------+----------------------+-------------+-------------------+----------------+-------------+-----------+----------+----------+------------+----------+----------+----------+------------------------------+---------+----------+------------+----------------+--------+----------+---------------+-----------------------------------------------+-----------+------+----------+-------------+----------------------------------+-----------+----------+----------------+--------------+---------------+-------------+-----------+--------------+----------------+---------------+--------+--------------+------------+-----------+--------------+----------------+-------------------+----------------+-----------------+---------------+-------------------+---------------+-------------------+---------+--------+-------------+--------------+---------------+-------------+------------+-------------+-------------+-----------+----------+------+----------+-----------+------------+--------------+------------+------------+--------------+--------------+---------+--------------+-----------------+------------------+-------------------------+----------+-----------+--------------------+---------------------+---------------------------+----------------+--------------+----------+------+------------+------------+-------------------------------------------+---------------------------------------------+-----------+-----------+------------+------------+ > | id | name | context | callingpres | deny > | permit | acl | secret | md5secret | remotesecret | transport | > host | nat | type | accountcode | amaflags | callgroup > | pickupgroup | namedcallgroup | namedpickupgroup | callerid | > directmedia | directmediapermit | directmediaacl | description | defaultip | > dtmfmode | fromuser | fromdomain | insecure | language | tonezone | mailbox > | qualify | regexten | rtptimeout | rtpholdtimeout | setvar | disallow | > allow | fullcontact | ipaddr | > port | username | defaultuser | dial | trustrpid > | sendrpid | progressinband | promiscredir | useclientcode | callcounter | > busylevel | allowoverlap | allowsubscribe | allowtransfer | lastms | > useragent | regseconds | regserver | videosupport | maxcallbitrate | > rfc2833compensate | session-timers | session-expires | session-minse | > session-refresher | outboundproxy | callbackextension | timert1 | timerb | > qualifyfreq | constantssrc | contactpermit | contactdeny | contactacl | > usereqphone | textsupport | faxdetect | buggymwi | auth | fullname | > trunkname | cid_number | mohinterpret | mohsuggest | parkinglot | > hasvoicemail | subscribemwi | vmexten | rtpkeepalive | g726nonstandard | > ignoresdpversion | subscribecontext | template | keepalive | > t38pt_usertpsource | organization_domain | outofcall_message_context | > sippasswd | kamailioname | mwi_from | avpf | dtlsenable | dtlsverify | > dtlscertfile | dtlsprivatekey > | dtlssetup | force_avp | icesupport | encryption | > +----+------------------------------+--------------------------------+-------------+-----------------+-----------------+------+--------+-----------+--------------+------------+---------+---------------------+--------+-------------+----------+-----------+-------------+----------------+------------------+----------------------+-------------+-------------------+----------------+-------------+-----------+----------+----------+------------+----------+----------+----------+------------------------------+---------+----------+------------+----------------+--------+----------+---------------+-----------------------------------------------+-----------+------+----------+-------------+----------------------------------+-----------+----------+----------------+--------------+---------------+-------------+-----------+--------------+----------------+---------------+--------+--------------+------------+-----------+--------------+----------------+-------------------+----------------+-----------------+---------------+-------------------+---------------+-------------------+---------+--------+-------------+--------------+---------------+-------------+------------+-------------+-------------+-----------+----------+------+----------+-----------+------------+--------------+------------+------------+--------------+--------------+---------+--------------+-----------------+------------------+-------------------------+----------+-----------+--------------------+---------------------+---------------------------+----------------+--------------+----------+------+------------+------------+-------------------------------------------+---------------------------------------------+-----------+-----------+------------+------------+ > | 12 | avillacisIM_pbx.villacis.com | pbx.villacis.com-from-internal | NULL > | 0.0.0.0/0.0.0.0 | 0.0.0.0/0.0.0.0 | NULL | NULL | NULL | NULL > | ws,wss,udp | dynamic | force_rport,comedia | friend | NULL | NULL
This is your issue, you are allowing ws and wss as the transport. Because you are forwarding calls over UDP, that is the only protocol you should be allowing. > | NULL | NULL | NULL | NULL | device > <avillacisIM> | no | NULL | NULL | NULL > | NULL | auto | NULL | NULL | NULL | es | NULL > | 1...@pbx.villacis.com-default | no | NULL | 60 | > 300 | NULL | all | ulaw,alaw,gsm | > sip:uqcma3g6@192.168.3.2:59675^3Btransport=ws | 127.0.0.1 | 5060 | > | avillacisIM | SIP/avillacisIM_pbx.villacis.com | yes | no | > NULL | NULL | NULL | yes | NULL | no > | NULL | yes | 0 | SIP.js/0.6.2 | 1409346610 | > | yes | 384 | NULL | NULL | NULL > | NULL | NULL | NULL | NULL | > NULL | NULL | 60 | NULL | NULL | NULL | > NULL | NULL | NULL | yes | NULL | NULL | 101 > | NULL | NULL | NULL | NULL | NULL | NULL > | NULL | *97 | NULL | NULL | NULL > | pbx.villacis.com-im-sip | NULL | NULL | NULL | > pbx.villacis.com | pbx.villacis.com-im-sip | Avillacis12345 | > avillacisIM | NULL | yes | yes | no | > /etc/pki/tls/certs/localhost_asterisk.crt | > /etc/pki/tls/private/localhost_asterisk.key | actpass | yes | yes | > yes | > +----+------------------------------+--------------------------------+-------------+-----------------+-----------------+------+--------+-----------+--------------+------------+---------+---------------------+--------+-------------+----------+-----------+-------------+----------------+------------------+----------------------+-------------+-------------------+----------------+-------------+-----------+----------+----------+------------+----------+----------+----------+------------------------------+---------+----------+------------+----------------+--------+----------+---------------+-----------------------------------------------+-----------+------+----------+-------------+----------------------------------+-----------+----------+----------------+--------------+---------------+-------------+-----------+--------------+----------------+---------------+--------+--------------+------------+-----------+--------------+----------------+-------------------+----------------+-----------------+---------------+-------------------+---------------+-------------------+---------+--------+-------------+--------------+---------------+-------------+------------+-------------+-------------+-----------+----------+------+----------+-----------+------------+--------------+------------+------------+--------------+--------------+---------+--------------+-----------------+------------------+-------------------------+----------+-----------+--------------------+---------------------+---------------------------+----------------+--------------+----------+------+------------+------------+-------------------------------------------+---------------------------------------------+-----------+-----------+------------+------------+ > 1 row in set (0.00 sec) > > > [root@elx3 kamailio]# asterisk -rnx 'sip show peer > avillacisIM_pbx.villacis.com' > > > * Name : avillacisIM_pbx.villacis.com > Description : > Realtime peer: Yes, cached > Secret : <Not set> > MD5Secret : <Not set> > Remote Secret: <Not set> > Context : pbx.villacis.com-from-internal > Record On feature : automon > Record Off feature : automon > Subscr.Cont. : pbx.villacis.com-im-sip > Language : es > Tonezone : <Not set> > AMA flags : Unknown > Transfer mode: open > CallingPres : Presentation Allowed, Not Screened > Callgroup : > Pickupgroup : > Named Callgr : > Nam. Pickupgr: > MOH Suggest : > Mailbox : 1...@pbx.villacis.com-default > VM Extension : *97 > LastMsgsSent : 0/0 > Call limit : 2147483647 > Max forwards : 0 > Dynamic : Yes > Callerid : "101" <avillacisIM> > MaxCallBR : 384 kbps > Expire : 153 > Insecure : no > Force rport : Yes > Symmetric RTP: Yes > ACL : Yes > DirectMedACL : No > T.38 support : Yes > T.38 EC mode : FEC > T.38 MaxDtgrm: 4294967295 > DirectMedia : No > PromiscRedir : No > User=Phone : No > Video Support: Yes > Text Support : No > Ign SDP ver : No > Trust RPID : Yes > Send RPID : No > TrustIDOutbnd: Legacy > Subscriptions: Yes > Overlap dial : No > DTMFmode : auto > Timer T1 : 500 > Timer B : 32000 > ToHost : > Addr->IP : 127.0.0.1:5060 > Defaddr->IP : (null) > Prim.Transp. : WS > Allowed.Trsp : UDP,WS,WSS Again, see above, this should only be udp. > Def. Username: avillacisIM > SIP Options : (none) > Codecs : (gsm|ulaw|alaw) > Codec Order : (ulaw:20,alaw:20,gsm:20) > Auto-Framing : No > Status : Unmonitored > Useragent : SIP.js/0.6.2 > Reg. Contact : sip:uqcma3g6@192.168.3.2:59675;transport=ws > Qualify Freq : 60000 ms > Keepalive : 0 ms > Sess-Timers : Accept > Sess-Refresh : uas > Sess-Expires : 1800 secs > Min-Sess : 90 secs > RTP Engine : asterisk > Parkinglot : > Use Reason : No > Encryption : Yes > Ign.Lifetime : No > > I think the situation is because of the change of transport. How should this > be handled so that Asterisk stops trying to use websocket transport for the > signaling that came from the UDP port? > See my responses above. -- 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
