<div dir="ltr">Hi,<div><br></div><div>Thanks Daniel for your reply. </div><div><br></div><div>Sorry for having been a bit obscure, it is my intention to have all clients able to call each other, regardless of which ua client software they use. I think I've realized what's going on. My goal is to use rtpengine to bridge between rtp profiles when they are different. But according to sip.js instruction, I set up my clients in a way that Asterisk took the place of rtpengine and changed the rtp profiles along the way based on the realtime table values. That got me confused but now I know at least what the problem is so I can fix it. This setup works in a way that I can make calls between websocket and sip clients, but the problem with it is that I need different values in the realtime table, according to which rtp profile the client uses.</div>
<div><br></div><div>Doing this I made a wrong turn in my project, I'll need to have "universal" setup for each peer so the user can use a websocket client or a sip client to register and use an account. I'll still need to figure out which settings to use and which not to use, so the rtp gets handled by rtpengine, not Asterisk. But that's a question for the Asterisk list.</div>
<div><br></div><div><br></div><div><br></div><div>The problem about Asterisk setting the rtp profile as <span style="color:rgb(80,0,80);font-family:arial,sans-serif;font-size:12.727272033691406px">UDP/TLS/RTP/SAVPF</span> was fixed using a peer setting in the realtime table, now Asterisk accepts RTP/SAVPF I can have calls flowing as soon as I can get rtpengine to cooperate with me.</div>
<div><span style="color:rgb(80,0,80);font-family:arial,sans-serif;font-size:12.727272033691406px"><br></span></div><div><span style="color:rgb(80,0,80);font-family:arial,sans-serif;font-size:12.727272033691406px">I wonder, is there </span><span style="color:rgb(80,0,80);font-family:arial,sans-serif;font-size:12.727272033691406px">UDP/TLS/RTP/SAVPF handling in rtpengine/kamailio? I may have to add some kind of handling to this if I have to revert back to my previous settings.</span></div>
<div><br></div><div>cheers,</div><div>Olli</div></div><div class="gmail_extra"><br><br><div class="gmail_quote">2014-08-05 16:49 GMT+03:00 Daniel-Constantin Mierla <span dir="ltr"><<a href="mailto:miconda@gmail.com" target="_blank">miconda@gmail.com</a>></span>:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div class=""><br>
On 01/08/14 10:56, Olli Heiskanen wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hi,<br>
<br>
I got ahead with my setup, this post helped me much: <a href="http://forums.digium.com/viewtopic.php?f=1&t=90167&sid=66fdf8cc4be5d955ba584e989a23442f" target="_blank">http://forums.digium.com/<u></u>viewtopic.php?f=1&t=90167&sid=<u></u>66fdf8cc4be5d955ba584e989a2344<u></u>2f</a><br>
<br>
At least the avpf setting had to be removed from sip.conf and put in the realtime db table, defined per client. I left the encryption setting in sip.conf. I had some problems calling from SIP client to another, then had to define avpf=no for those clients. Personally I don't like to use different settings to different clients, is there a way around this?<br>
<br>
With this setup I can make calls between SIP clients but not ws clients. My client (now I use sip.js) fails to parse the sdp - including the apparently correct rtp profile UDP/TLS/RTP/SAVPF - and sends back 488, which makes the call fail. I'd like to hear opinions from you guys which would be the correct place to handle this? My setup has Asterisk Kamailio realtime integration, and I use dispatcher in Kamailio to route calls to Asterisk. Kamailio sounds like the logical place, but I'd rather find a way to not change the rtp profile along the way, at least until the clients can support that one.<br>
</blockquote></div>
To understand properly, you don't want to use rtpenging for srtp(webrtc)-rtp(classic sip) gatewaying?<br>
<br>
If yes, maybe you can partition the users (classic-sip and webrtc-sip), then use two asterisk instances with routing via kamailio.<br>
<br>
Cheers,<br>
Daniel<span class="HOEnZb"><font color="#888888"><br>
<br>
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