<html><head><meta http-equiv="Content-Type" content="text/html charset=utf-8"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class="">The only caveat to this is, if you are registering to freeswitch and use that registration to send the browser a call, it will not know to send an sdp that will work with webrtc. You will need to tell it to do so using media_webrtc=true var on originate.<div class=""><br class=""><div><blockquote type="cite" class=""><div class="">On Jan 10, 2015, at 1:38 PM, Adam Ben-Ayoun <<a href="mailto:adam.ben.ayoun1@gmail.com" class="">adam.ben.ayoun1@gmail.com</a>> wrote:</div><br class="Apple-interchange-newline"><div class=""><div dir="ltr" class="">Great to hear that. Thanks again.</div><div class="gmail_extra"><br class=""><div class="gmail_quote">On 10 January 2015 at 20:35, Carlos Ruiz Díaz <span dir="ltr" class=""><<a href="mailto:carlos.ruizdiaz@gmail.com" target="_blank" class="">carlos.ruizdiaz@gmail.com</a>></span> wrote:<br class=""><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><p dir="ltr" class="">WebRTC doesn't specify a signalling protocol. This means that you can use SIP over any transport you want to carry the webRTC enabled SDP. </p><p dir="ltr" class="">FS will receive the SDP, detect that has a RTP/SAVPF profile and start handling it accordingly.</p><p dir="ltr" class="">Take for example Jitsi or IMSDroid, they both support webRTC and do SIP over UDP/TCP/TLS. </p><p dir="ltr" class="">Regards, <br class="">
Carlos </p><div class="HOEnZb"><div class="h5">
<div class="gmail_quote">On Jan 10, 2015 12:08 PM, "Adam Ben-Ayoun" <<a href="mailto:adam.ben.ayoun1@gmail.com" target="_blank" class="">adam.ben.ayoun1@gmail.com</a>> wrote:<br type="attribution" class=""><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr" class="">Thanks Anthony. I assume that means I can use SIP over TCP/TLS for signalling? Also, will mandatory WebRTC requirements such as DTLS-SRTP work when communicating with FS (when stuff like fingerprint, etc)?</div><div class="gmail_extra"><br class=""><div class="gmail_quote">On 10 January 2015 at 19:49, Anthony Minessale <span dir="ltr" class=""><<a href="mailto:anthony.minessale@gmail.com" target="_blank" class="">anthony.minessale@gmail.com</a>></span> wrote:<br class=""><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr" class="">The WebRTC media engine is driven completely by the SDP, the transport will not make any difference.<div class=""><br class=""></div></div><div class="gmail_extra"><br class=""><div class="gmail_quote"><div class=""><div class="">On Fri, Jan 9, 2015 at 5:26 PM, Adam Ben-Ayoun <span dir="ltr" class=""><<a href="mailto:adam.ben.ayoun1@gmail.com" target="_blank" class="">adam.ben.ayoun1@gmail.com</a>></span> wrote:<br class=""></div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div class=""><div class=""><div dir="ltr" class="">Hi,<div class=""><br class=""></div><div class=""><div class="">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 class=""><br class=""></div><div class="">Thanks,</div><div class="">Adam</div></div>
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