<html><head><meta http-equiv="Content-Type" content="text/html charset=windows-1252"><base href="x-msg://46123/"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">why would you set sip_secure_media?<div><br><div><div>On Jun 26, 2013, at 10:41 AM, Gerald Weber <<a href="mailto:gerald.weber@besharp.at">gerald.weber@besharp.at</a>> wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><div lang="DE-AT" link="blue" vlink="purple" style="font-family: Helvetica; font-size: medium; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-align: -webkit-auto; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; "><div class="WordSection1" style="page: WordSection1; "><div style="margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); ">Hi,<o:p></o:p></span></div><div style="margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); "> </span></div><div style="margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); ">how did you get audio working ?<o:p></o:p></span></div><div style="margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); ">Ive also tried jssip and latest chrome and sip works, but there is no audio. Codecs like opus, ilbc and isac are installed.<o:p></o:p></span></div><div style="margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); ">In the dialplan i set sip_secure_media=true but the speaker stays silent. Ive tried ws and wss bindings but no luck.<o:p></o:p></span></div><div style="margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); "> </span></div><div style="margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); ">Thanks for any suggestions,<o:p></o:p></span></div><div style="margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); ">gw<o:p></o:p></span></div><div style="margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); "> </span></div><div style="margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "><b><span lang="DE" style="font-size: 10pt; font-family: Tahoma, sans-serif; ">Von:</span></b><span lang="DE" style="font-size: 10pt; font-family: Tahoma, sans-serif; "><span class="Apple-converted-space"> </span><a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a> [mailto:freeswitch-<a href="mailto:users-bounces@lists.freeswitch.org">users-bounces@lists.freeswitch.org</a>]<span class="Apple-converted-space"> </span><b>Im Auftrag von<span class="Apple-converted-space"> </span></b>Joćo Mesquita<br><b>Gesendet:</b><span class="Apple-converted-space"> </span>Mittwoch, 26. Juni 2013 15:28<br><b>An:</b><span class="Apple-converted-space"> </span>FreeSWITCH Users Help<br><b>Betreff:</b><span class="Apple-converted-space"> </span>Re: [Freeswitch-users] FreeSWITCH adds WebRTC support to new 1.4 BETA.<o:p></o:p></span></div><div style="margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "><o:p> </o:p></div><div><div style="margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; ">Whooray!!!! I've installed the new beta yesterday and tried it out with Chrome+JsSIP and it works fantastic!!!<o:p></o:p></div><div><div style="margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "><o:p> </o:p></div></div><div><div style="margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; ">One question tho, is it possible/desirable to have a sofia profile to work only with webrtc and not with udp or tcp bindings?<o:p></o:p></div></div><div><div style="margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "><o:p> </o:p></div></div><div><div style="margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; ">I haven't tried and I guess I should've before asking dumb questions but....<o:p></o:p></div></div></div><div><div style="margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "><br clear="all"><o:p></o:p></div><div><div style="margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; ">Joćo Mesquita<br>FreeSWITCH Solutions<o:p></o:p></div></div><p class="MsoNormal" style="margin: 0cm 0cm 12pt; font-size: 12pt; font-family: 'Times New Roman', serif; "><o:p> </o:p></p><div><div style="margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; ">On Tue, Jun 25, 2013 at 7:42 PM, Anthony Minessale <<a href="mailto:anthony.minessale@gmail.com" target="_blank" style="color: purple; text-decoration: underline; ">anthony.minessale@gmail.com</a>> wrote:<o:p></o:p></div><div><div style="margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; ">Now also working in latest FireFox 22 <o:p></o:p></div><div><div style="margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "><o:p> </o:p></div></div><div><div style="margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; ">sipml5 works with both chrome and FF<o:p></o:p></div></div><div><div style="margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "><o:p> </o:p></div></div><div><div style="margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "><a href="https://webrtc.freeswitch.org/sipml5/" target="_blank" style="color: purple; text-decoration: underline; ">https://webrtc.freeswitch.org/sipml5/</a><o:p></o:p></div></div><div><div style="margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "><o:p> </o:p></div></div></div><div><p class="MsoNormal" style="margin: 0cm 0cm 12pt; font-size: 12pt; font-family: 'Times New Roman', serif; "><o:p> </o:p></p><div><div style="margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; ">On Tue, Jun 25, 2013 at 2:04 AM, Michael Jerris <<a href="mailto:mike@jerris.com" target="_blank" style="color: purple; text-decoration: underline; ">mike@jerris.com</a>> wrote:<o:p></o:p></div><div style="margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; ">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<o:p></o:p></div><div><div style="margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "><br>On Jun 25, 2013, at 12:37 AM, Jeff Leung <<a href="mailto:jleung@v10networks.ca" target="_blank" style="color: purple; text-decoration: underline; ">jleung@v10networks.ca</a>> wrote:<br><br>> Media wise, how would that be handled with web sockets?<br>><br>>> -----Original Message-----<br>>> From:<span class="Apple-converted-space"> </span><a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank" style="color: purple; text-decoration: underline; ">freeswitch-users-bounces@lists.freeswitch.org</a><span class="Apple-converted-space"> </span>[mailto:<a href="mailto:freeswitch-" target="_blank" style="color: purple; text-decoration: underline; ">freeswitch-</a><br>>><span class="Apple-converted-space"> </span><a href="mailto:users-bounces@lists.freeswitch.org" target="_blank" style="color: purple; text-decoration: underline; ">users-bounces@lists.freeswitch.org</a>] On Behalf Of Dan Barua<br>>> Sent: Monday, June 24, 2013 3:10 PM<br>>> To:<span class="Apple-converted-space"> </span><a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank" style="color: purple; text-decoration: underline; ">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></div></div></div></div></div></div></div></div></blockquote></div></div></body></html>