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