[Freeswitch-users] SIP Phone Integration

[Madovsky]

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 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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