[Freeswitch-users] SIP Phone Integration

[David Ponzone]

If you choose another codec, you could use a HW-based transcoding solution like the Sangoma D100/500, which goes from 30 to 2000 concurrent calls.
According their website, they don't support Speex.

David Ponzone  Direction Technique
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Le 06/12/2010 à 16:23, James Mbuthia a écrit :

> Thanks,  about how many concurrent calls are you able to when transcoding with FS?
> 
> On Mon, Dec 6, 2010 at 5:17 PM, Madovsky <infos at madovsky.org> wrote:
> You can transcode with FS,
> I did already a web phone (boophone) with FS but ;keep in mind
> that transcode=CPU
> ----- Original Message -----
> From: James Mbuthia
> To: freeswitch-users at lists.freeswitch.org
> Sent: Monday, December 06, 2010 9:49 AM
> Subject: [Freeswitch-users] SIP Phone Integration
> 
> Hi guys,
> 
> Am new to Freeswitch and am looking for info that can help me develop a web-based SIP Phone
> 
>  I am developing a web-based SIP Phone based on PHP/MySQL and OpenSIPS. I have developed the component of the SIP phone responsible for the 3way handshake and SDP offer. 
> 
> My challenge is to now integrate the application to a rtp stack which will enable the app to pick up audio and transmit it over the internet. Ultimately I want to connect the app to the PSTN using a media server such as Freeswitch or Asterisk. I want to use the Speex codec. I wanted to know whether Freeswitch has a rtp stack integrated with speex and whether there any tutorials on how integration to a softphone can be done. Any pointers or ideas from you would be very helpful and highly appreaciated.
> 
> regards,
> James Mbuthia
> 
> 
> 
> 
> 
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