<div dir="ltr">Did you use the api from scratch or did you use their example site as base? I tried with the bare API and had the same behavior as you did, but after I tried with the example as base and it worked, I assumed that I was doing something wrong.<div>
<br></div><div style>Also, JsSip has 0.3 and 0.2, I am working with 0.2 and it seems that 0.3 has API changes...</div></div><div class="gmail_extra"><br clear="all"><div>Joćo Mesquita<br>FreeSWITCH Solutions<br></div>
<br><br><div class="gmail_quote">On Wed, Jun 26, 2013 at 11:41 AM, Gerald Weber <span dir="ltr"><<a href="mailto:gerald.weber@besharp.at" target="_blank">gerald.weber@besharp.at</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div lang="DE-AT" link="blue" vlink="purple"><div><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Hi,<u></u><u></u></span></p><p class="MsoNormal">
<span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">how did you get audio working ?<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Ive also tried jssip and latest chrome and sip works, but there is no audio. Codecs like opus, ilbc and isac are installed.<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">In the dialplan i set sip_secure_media=true but the speaker stays silent. Ive tried ws and wss bindings but no luck.<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Thanks for any suggestions,<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">gw<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p>
<p class="MsoNormal"><b><span lang="DE" style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">Von:</span></b><span lang="DE" style="font-size:10.0pt;font-family:"Tahoma","sans-serif""> <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a> [mailto:<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a>] <b>Im Auftrag von </b>Joćo Mesquita<br>
<b>Gesendet:</b> Mittwoch, 26. Juni 2013 15:28<br><b>An:</b> FreeSWITCH Users Help<br><b>Betreff:</b> Re: [Freeswitch-users] FreeSWITCH adds WebRTC support to new 1.4 BETA.<u></u><u></u></span></p><div><div class="h5"><p class="MsoNormal">
<u></u> <u></u></p><div><p class="MsoNormal">Whooray!!!! I've installed the new beta yesterday and tried it out with Chrome+JsSIP and it works fantastic!!!<u></u><u></u></p><div><p class="MsoNormal"><u></u> <u></u></p>
</div><div><p class="MsoNormal">One question tho, is it possible/desirable to have a sofia profile to work only with webrtc and not with udp or tcp bindings?<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p>
</div><div><p class="MsoNormal">I haven't tried and I guess I should've before asking dumb questions but....<u></u><u></u></p></div></div><div><p class="MsoNormal"><br clear="all"><u></u><u></u></p><div><p class="MsoNormal">
Joćo Mesquita<br>FreeSWITCH Solutions<u></u><u></u></p></div><p class="MsoNormal" style="margin-bottom:12.0pt"><u></u> <u></u></p><div><p class="MsoNormal">On Tue, Jun 25, 2013 at 7:42 PM, Anthony Minessale <<a href="mailto:anthony.minessale@gmail.com" target="_blank">anthony.minessale@gmail.com</a>> wrote:<u></u><u></u></p>
<div><p class="MsoNormal">Now also working in latest FireFox 22 <u></u><u></u></p><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">sipml5 works with both chrome and FF<u></u><u></u></p></div><div>
<p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal"><a href="https://webrtc.freeswitch.org/sipml5/" target="_blank">https://webrtc.freeswitch.org/sipml5/</a><u></u><u></u></p></div><div><p class="MsoNormal">
<u></u> <u></u></p></div></div><div><div><div><p class="MsoNormal" style="margin-bottom:12.0pt"><u></u> <u></u></p><div><p class="MsoNormal">On Tue, Jun 25, 2013 at 2:04 AM, Michael Jerris <<a href="mailto:mike@jerris.com" target="_blank">mike@jerris.com</a>> wrote:<u></u><u></u></p>
<p class="MsoNormal">Media is not handled by websockets. Web-sockets are a new transport for sip signaling. If an endpoint registers from a websocket transport, when we send it a call, we automatically send it an sdp appropriate for webrtc. If you want to send a call with an sdp for webrtc that isn't for a registration over websockets, just add media_webrtc=true var to the origination vars. If we receive an sdp from webrtc, we will automatically handle it.<br>
<br>Mike<u></u><u></u></p><div><div><p class="MsoNormal"><br>On Jun 25, 2013, at 12:37 AM, Jeff Leung <<a href="mailto:jleung@v10networks.ca" target="_blank">jleung@v10networks.ca</a>> wrote:<br><br>> Media wise, how would that be handled with web sockets?<br>
><br>>> -----Original Message-----<br>>> From: <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a> [mailto:<a href="mailto:freeswitch-" target="_blank">freeswitch-</a><br>
>> <a href="mailto:users-bounces@lists.freeswitch.org" target="_blank">users-bounces@lists.freeswitch.org</a>] On Behalf Of Dan Barua<br>>> Sent: Monday, June 24, 2013 3:10 PM<br>>> To: <a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a><br>
>> Subject: Re: [Freeswitch-users] FreeSWITCH adds WebRTC support to new<br>>> 1.4 BETA.<br>>><br>>> I asked the same question on Google+, Michael Jerris helpfully replied:<br>>><br>>> you just need to setup FreeSWITCH as normal, and add 2 configuration<br>
>> params for the binding addresses for the websockets and secure<br>>> websockets. the params are ws-binding and wss-binding format is ip:port or<br>>> just :port if your using the same ip as sip-ip<br>
>><br>>><br>>> I haven't got round to trying it myself yet.<br>>><br>>> -Dan<br>>><br>>><br>>><br>>><br>>> --<br>>> View this message in context: <a href="http://freeswitch-" target="_blank">http://freeswitch-</a><br>
>> <a href="http://users.2379917.n2.nabble.com/FreeSWITCH-adds-WebRTC-support-to-new-" target="_blank">users.2379917.n2.nabble.com/FreeSWITCH-adds-WebRTC-support-to-new-</a><br>>> 1-4-BETA-tp7592138p7592141.html<br>
>> Sent from the freeswitch-users mailing list archive at Nabble.com.<br>>><br>>> __________________________________________________________<br>>> _______________<br>>> Professional FreeSWITCH Consulting Services:<br>
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</div></div></div><div><p class="MsoNormal">-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/" target="_blank">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/" target="_blank">http://www.cluecon.com/</a><br>
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<br></blockquote></div><br></div>