Okay, tried reverting to Asterisk 11.10.2. I didn't change the realtime table yet, but now when calling from websocket client to another websocket client, cli says:
WARNING[30620][C-00000000]: chan_sip.c:11056 process_sdp_a_dtls: Unsupported fingerprint hash type 'sha-2' received on dialog '36ns50nk1fo04pu3m7lf' WARNING[30620][C-00000000]: chan_sip.c:10509 process_sdp: Rejecting secure audio stream without encryption details: audio 10640 RTP/SAVPF 111 103 104 0 8 106 105 13 126 This many times, until the forking capacity of Kamailio has been reached and call fails. The clients are running on chrome, and calls have worked before... I wonder if I should revert further back and/or change or remove some realtime table fields? cheers, Olli 2014-08-12 11:17 GMT+03:00 Olli Heiskanen <[email protected]>: > Hello, > > Thank You Paul for your reply, > > The registrations in my setup are not duplicated, the 'secret' field in > the realtime table is empty, which causes Asterisk to not authenticate > requests from my Kamailio. Kamailio handles registrations, and also routes > the traffic to Asterisk using dispatcher. Also, all peers have the Kamailio > ip:port as outbound proxy so all traffic goes through Kamailio. > > Looks like version 11.11 works differently, I'll try to revert back to a > previous version, and see if that works. I know at least the 'force_avp' > field is new to 11.11 so it's safe to assume there's some difference > between versions in rtp profile handling. > > It would be good to know how to handle this scenario in the new versions > as well, I'll probably need to upgrade ahead anyway. > > Thanks, > Olli > > > > 2014-08-12 1:56 GMT+03:00 Paul Belanger <[email protected]>: > > On Mon, Aug 11, 2014 at 4:45 AM, Olli Heiskanen >> <[email protected]> wrote: >> > >> > Hello, >> > >> > I'm trying to get calls working between websocket clients and sip >> clients. >> > For clients I have sip.js based clients on chrome, Zoipers and a >> Grandstream >> > phone. Challenge here is I'd like to have Kamailio and rtpengine to >> handle >> > the bridging between different rtp profiles but Asterisk changes them >> in the >> > sdp bodies along the way. I'm using Asterisk 11.11.0. >> > >> > Is there a way to configure Asterisk to ignore the rtp profile but allow >> > calls to pass with either of those profiles (even though clients might >> > answer with 488 which would be caught and handled by Kamailio and >> > rtpengine)? In my setup I have Asterisk Kamailio realtime integration, >> and >> > the second goal is to be able to add peers to the db table with similar >> > data, as in no different values based on what kind of client wants to >> > register. I'd like to allow the user to register using which ever client >> > they choose (in this case one of the 3 I mentioned). >> > >> > Previously I had problems like 'rejecting secure audio stream without >> > encryption details', no audio or BYE messages sent immediately after >> call >> > has begun etc, but according to sip.js documentation >> > (http://sipjs.com/guides/server-configuration/asterisk/) the settings >> avpf >> > and force_avp affect the way Asterisk handles the rtp profiles and now >> my >> > calls do work ok but I'd need to move the rtp profile handling to >> rtpengine. >> > >> We are successfully using kamailio / rtpengine with websockets and >> asterisk 1.8. First question is why are you duplicating registrations >> within asterisk? Secondly, why are you using websockets in asterisk? >> >> Without knowing more about your use case, I'll tell you how we did it. >> Like I said, kamailio is responsible for our SIP/ws subscribers and >> registrations. Once within kamailio we simply dispatch traffic to >> asterisk via SIP/udp. RTP is handled by rtpengine (using rtproxy-ng) >> and that is basically it. >> >> No special configuration is needed for asterisk (in fact 1.8 has no >> support for RTP/SAVPF) so we rewrite SDP on 488. Then setup a >> kamailio peer and away you go. >> >> -- >> Paul Belanger | PolyBeacon, Inc. >> Jabber: [email protected] | IRC: pabelanger (Freenode) >> Github: https://github.com/pabelanger | Twitter: >> https://twitter.com/pabelanger >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > >
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