+1 as well.<br><br>Good value if we want Opensips to be a drop in replacement for SBC <br>combine with the work that was done on B2BUA.<br><br>I suppose that using Sangoma we will have very little latency introduced on the media.<br>
<br><br><div class="gmail_quote">On Mon, Oct 29, 2012 at 7:06 AM, SamyGo <span dir="ltr"><<a href="mailto:govoiper@gmail.com" target="_blank">govoiper@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
+1for <span style="line-height:21.58333396911621px;color:rgb(102,102,102);font-size:13px;font-family:'Trebuchet MS',sans-serif,Verdana,Arial">new module to drive SANGOMA cards in for transcoding (similar to driving the rtpproxy or mediaproxy)</span><br>
<br><div>Regards,<div>Sammy<br><br><div class="gmail_quote"><div><div class="h5">On Sat, Oct 27, 2012 at 4:41 AM, Ali Pey <span dir="ltr"><<a href="mailto:alipey@gmail.com" target="_blank">alipey@gmail.com</a>></span> wrote:<br>
</div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div class="h5">
I do also see a lot of value in sip over websocket. WebRTC is pretty much here and it makes much more sense to be able to support it on one proxy server rather than having to use OverSIP and then OpenSIPS. WebRTC seems to be very popular and the thing of tomorrow and it will be very important for a sip proxy server to support it.<div>
<br></div><div>Regards,</div><div>Ali Pey<br><br><div class="gmail_quote"><div>On Fri, Oct 26, 2012 at 11:54 AM, Saúl Ibarra Corretgé <span dir="ltr"><<a href="mailto:saul@ag-projects.com" target="_blank">saul@ag-projects.com</a>></span> wrote:<br>
</div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div><br>
On Oct 26, 2012, at 5:46 PM, Duane Larson wrote:<br>
<br>
> Is there any roadmap for "SIP over Websocket"? I know there is now OverSIP but wasn't sure if OpenSIPS had any plans to implement a module. Just asking since WebRTC is still evolving.<br>
><br>
<br>
</div>I personally don't see the need to do it in the 1.x series. You can use OverSIP to do protocol translation to TCP/TLS and send the call to OpenSIPS. Now, IIRC there was some problem in parsing the Via headers, because they now have a different transport parameter (ws and wss), but I think someone posted a patch already.<br>
<div><br>
--<br>
Saúl Ibarra Corretgé<br>
AG Projects<br>
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