[Freeswitch-users] SIP Phone Integration

[James Mbuthia]

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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