[Freeswitch-users] SIP Phone Integration

[James Mbuthia]

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 <jmmbuthia at gmail.com>
> *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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