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. Also, JsSip has 0.3 and 0.2, I am working with 0.2 and it seems that 0.3 has API changes... João Mesquita FreeSWITCH™ Solutions On Wed, Jun 26, 2013 at 11:41 AM, Gerald Weber <gerald.weber@besharp.at>wrote:
Hi,****
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how did you get audio working ?****
I’ve also tried jssip and latest chrome and sip works, but there is no audio. Codecs like opus, ilbc and isac are installed.****
In the dialplan i set sip_secure_media=true but the speaker stays silent. I’ve tried ws and wss bindings but no luck.****
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Thanks for any suggestions,****
gw****
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*Von:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *Im Auftrag von *João Mesquita *Gesendet:* Mittwoch, 26. Juni 2013 15:28 *An:* FreeSWITCH Users Help *Betreff:* Re: [Freeswitch-users] FreeSWITCH adds WebRTC support to new 1.4 BETA.****
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Whooray!!!! I've installed the new beta yesterday and tried it out with Chrome+JsSIP and it works fantastic!!!****
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One question tho, is it possible/desirable to have a sofia profile to work only with webrtc and not with udp or tcp bindings?****
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I haven't tried and I guess I should've before asking dumb questions but....****
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João Mesquita FreeSWITCH™ Solutions****
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On Tue, Jun 25, 2013 at 7:42 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:****
Now also working in latest FireFox 22 ****
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sipml5 works with both chrome and FF****
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https://webrtc.freeswitch.org/sipml5/****
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On Tue, Jun 25, 2013 at 2:04 AM, Michael Jerris <mike@jerris.com> wrote:** **
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.
Mike****
On Jun 25, 2013, at 12:37 AM, Jeff Leung <jleung@v10networks.ca> wrote:
Media wise, how would that be handled with web sockets?
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch- users-bounces@lists.freeswitch.org] On Behalf Of Dan Barua Sent: Monday, June 24, 2013 3:10 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] FreeSWITCH adds WebRTC support to new 1.4 BETA.
I asked the same question on Google+, Michael Jerris helpfully replied:
you just need to setup FreeSWITCH as normal, and add 2 configuration params for the binding addresses for the websockets and secure websockets. the params are ws-binding and wss-binding format is ip:port or just :port if your using the same ip as sip-ip
I haven't got round to trying it myself yet.
-Dan
-- View this message in context: http://freeswitch- users.2379917.n2.nabble.com/FreeSWITCH-adds-WebRTC-support-to-new- 1-4-BETA-tp7592138p7592141.html Sent from the freeswitch-users mailing list archive at Nabble.com.
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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