<div dir="ltr"><br><div>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:</div><div><br></div><div>WARNING[30620][C-00000000]: chan_sip.c:11056 process_sdp_a_dtls: Unsupported fingerprint hash type 'sha-2' received on dialog '36ns50nk1fo04pu3m7lf'<br>
</div><div>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<br></div><div><br></div><div>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?</div>
<div><br></div><div>cheers,</div><div>Olli</div></div><div class="gmail_extra"><br><br><div class="gmail_quote">2014-08-12 11:17 GMT+03:00 Olli Heiskanen <span dir="ltr"><<a href="mailto:ohjelmistoarkkitehti@gmail.com" target="_blank">ohjelmistoarkkitehti@gmail.com</a>></span>:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Hello,<div><br></div><div>Thank You Paul for your reply,</div><div><br></div><div>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.</div>
<div><br></div><div>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. </div>
<div><br></div><div>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.</div><div><br></div><div>Thanks,</div><div>Olli</div><div><br></div></div>
<div class="gmail_extra"><br><br><div class="gmail_quote">2014-08-12 1:56 GMT+03:00 Paul Belanger <span dir="ltr"><<a href="mailto:paul.belanger@polybeacon.com" target="_blank">paul.belanger@polybeacon.com</a>></span>:<div>
<div class="h5"><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div>On Mon, Aug 11, 2014 at 4:45 AM, Olli Heiskanen<br>
<<a href="mailto:ohjelmistoarkkitehti@gmail.com" target="_blank">ohjelmistoarkkitehti@gmail.com</a>> wrote:<br>
><br>
> Hello,<br>
><br>
> I'm trying to get calls working between websocket clients and sip clients.<br>
> For clients I have sip.js based clients on chrome, Zoipers and a Grandstream<br>
> phone. Challenge here is I'd like to have Kamailio and rtpengine to handle<br>
> the bridging between different rtp profiles but Asterisk changes them in the<br>
> sdp bodies along the way. I'm using Asterisk 11.11.0.<br>
><br>
> Is there a way to configure Asterisk to ignore the rtp profile but allow<br>
> calls to pass with either of those profiles (even though clients might<br>
> answer with 488 which would be caught and handled by Kamailio and<br>
> rtpengine)? In my setup I have Asterisk Kamailio realtime integration, and<br>
> the second goal is to be able to add peers to the db table with similar<br>
> data, as in no different values based on what kind of client wants to<br>
> register. I'd like to allow the user to register using which ever client<br>
> they choose (in this case one of the 3 I mentioned).<br>
><br>
> Previously I had problems like 'rejecting secure audio stream without<br>
> encryption details', no audio or BYE messages sent immediately after call<br>
> has begun etc, but according to sip.js documentation<br>
> (<a href="http://sipjs.com/guides/server-configuration/asterisk/" target="_blank">http://sipjs.com/guides/server-configuration/asterisk/</a>) the settings avpf<br>
> and force_avp affect the way Asterisk handles the rtp profiles and now my<br>
> calls do work ok but I'd need to move the rtp profile handling to rtpengine.<br>
><br>
</div>We are successfully using kamailio / rtpengine with websockets and<br>
asterisk 1.8. First question is why are you duplicating registrations<br>
within asterisk? Secondly, why are you using websockets in asterisk?<br>
<br>
Without knowing more about your use case, I'll tell you how we did it.<br>
Like I said, kamailio is responsible for our SIP/ws subscribers and<br>
registrations. Once within kamailio we simply dispatch traffic to<br>
asterisk via SIP/udp. RTP is handled by rtpengine (using rtproxy-ng)<br>
and that is basically it.<br>
<br>
No special configuration is needed for asterisk (in fact 1.8 has no<br>
support for RTP/SAVPF) so we rewrite SDP on 488. Then setup a<br>
kamailio peer and away you go.<br>
<span><font color="#888888"><br>
--<br>
Paul Belanger | PolyBeacon, Inc.<br>
Jabber: <a href="mailto:paul.belanger@polybeacon.com" target="_blank">paul.belanger@polybeacon.com</a> | IRC: pabelanger (Freenode)<br>
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