or use acme company,

but very expensive

----- Original Message -----

From: David Ponzone

To: FreeSWITCH Users Help

Sent: Monday, December 06, 2010 12:56 PM

Subject: Re: [Freeswitch-users] SIP Phone Integration

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

email: david.ponzone@ipeva.fr

tel:      01 74 03 18 97

gsm:   06 66 98 76 34

Service Client IPeva

tel:      0811 46 26 26

www.ipeva.fr  -   www.ipeva-studio.com

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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@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@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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