<div dir="ltr">The WebRTC media engine is driven completely by the SDP, the transport will not make any difference.<div><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Fri, Jan 9, 2015 at 5:26 PM, Adam Ben-Ayoun <span dir="ltr"><<a href="mailto:adam.ben.ayoun1@gmail.com" target="_blank">adam.ben.ayoun1@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Hi,<div><br></div><div><div>We are developing a mobile client that will use the WebRTC media stack and Freeswitch as an MCU (only for conference calls). My question is, since we build a native app, can we use SIP over TCP for signalling? In other words, if Freeswitch receives the WebRTC kind of SDP, will it be able to communicate in the same way as if we were using the SIP over Websocket (the other Freeswitch option)? Any corner cases/considerations with this? Our goal is to avoid implementing SIP over Websocket on the client as much as possible.</div></div><div><br></div><div>Thanks,</div><div>Adam</div></div>
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<a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br></blockquote></div><br><br clear="all"><div><br></div>-- <br><div class="gmail_signature"><div dir="ltr">Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬<div><br><div>☞ <a href="http://freeswitch.org/" target="_blank">http://freeswitch.org/</a> ☞ <a href="http://cluecon.com/" target="_blank">http://cluecon.com/</a> ☞ <a href="http://twitter.com/FreeSWITCH" target="_blank">http://twitter.com/FreeSWITCH</a></div><div><div>☞ <a href="http://irc.freenode.net" target="_blank">irc.freenode.net</a> #freeswitch ☞ <u><a href="http://freeswitch.org/g+" target="_blank">http://freeswitch.org/g+</a></u><br><br></div><div>ClueCon Weekly Development Call <br></div><div>☎ <a href="mailto:sip%3A888@conference.freeswitch.org" target="_blank">sip:888@conference.freeswitch.org</a> ☎ +19193869900 </div><div><br></div></div></div></div></div>
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