[Freeswitch-users] WebRTC Call

[Muhammad Shahzad]

Also you can use is WebRTC2SIP gateway which can connect to any traditional
SIP server, like freeswitch or asterisk.

http://code.google.com/p/webrtc2sip/

Thank you.


On Tue, Dec 4, 2012 at 11:43 AM, qasimakhan at gmail.com
<qasimakhan at gmail.com>wrote:

> I have successfuly made SIP-SIP (Both Audio & Video) calls using OverSIPs
> as WebRTC-SIP gateway and using Opensips 1.7 with VIA patch for WebRTC.
>
> Regards,
> Qasim
>
> On Sat, Dec 1, 2012 at 2:34 AM, Ros P <rosphotos82 at gmail.com> wrote:
>
>> Hello,
>>
>> Has anyone had any luck in implementing a webrtc call? I have been able
>> to register using the webrtc2sip gw and sipml5 client but no audio goes
>> through. As this seems to be a client problem (issue 39<http://code.google.com/p/sipml5/issues/detail?id=39>),
>> does anyone have any other suggestions? Anything successfully tested?
>>
>> Kind Regards,
>> ros
>>
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>
> _________________________________________________________________________
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> http://www.freeswitchsolutions.com
>
> 
> 
>
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>
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-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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