[Freeswitch-users] FreeSWITCH adds WebRTC support to new 1.4 BETA.

Thu Jun 27 18:07:08 MSD 2013

Ps: i also included the output from jssip in the pastebin link

Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Anthony Minessale
Gesendet: Donnerstag, 27. Juni 2013 15:16
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] FreeSWITCH adds WebRTC support to new 1.4 BETA.

try adding the sofia profile param

apply-candidate-acl=localnet.auto

On Thu, Jun 27, 2013 at 4:45 AM, Gerald Weber <gerald.weber at besharp.at<mailto:gerald.weber at besharp.at>> wrote:
I got rid of sip_secure_media and tried again, here is the trace:
http://pastebin.freeswitch.org/21116

SIP User 2020 on Chrome calls Ext 20000 (the funny prompts thing)
I'm no expert, but i guess there is something wrong with this stun/run/ice thing...

Thanks,
gw

Von: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] Im Auftrag von Anthony Minessale
Gesendet: Mittwoch, 26. Juni 2013 17:24

An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] FreeSWITCH adds WebRTC support to new 1.4 BETA.

You don't have to set anything in the dialplan.
Get a debug console trace and post it to pastebin.

On Wed, Jun 26, 2013 at 9:41 AM, Gerald Weber <gerald.weber at besharp.at<mailto:gerald.weber at besharp.at>> wrote:
Hi,

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.

Thanks for any suggestions,
gw

Von: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at 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.

Whooray!!!! I've installed the new beta yesterday and tried it out with Chrome+JsSIP and it works fantastic!!!

One question tho, is it possible/desirable to have a sofia profile to work only with webrtc and not with udp or tcp bindings?

I haven't tried and I guess I should've before asking dumb questions but....

João Mesquita
FreeSWITCH(tm) Solutions

On Tue, Jun 25, 2013 at 7:42 PM, Anthony Minessale <anthony.minessale at gmail.com<mailto:anthony.minessale at gmail.com>> wrote:
Now also working in latest FireFox 22

sipml5 works with both chrome and FF

https://webrtc.freeswitch.org/sipml5/

On Tue, Jun 25, 2013 at 2:04 AM, Michael Jerris <mike at jerris.com<mailto:mike at 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 at v10networks.ca<mailto:jleung at v10networks.ca>> wrote:

> Media wise, how would that be handled with web sockets?
>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-<mailto:freeswitch->
>> users-bounces at lists.freeswitch.org<mailto:users-bounces at lists.freeswitch.org>] On Behalf Of Dan Barua
>> Sent: Monday, June 24, 2013 3:10 PM
>> To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at 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-<http://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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_________________________________________________________________________
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--
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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
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Official FreeSWITCH Sites
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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
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--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com<mailto:MSN%3Aanthony_minessale at hotmail.com>
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FreeSWITCH Developer Conference
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pstn:+19193869900

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com

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--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com<mailto:MSN%3Aanthony_minessale at hotmail.com>
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