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 <jmmbuthia@gmail.com> *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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