[Freeswitch-users] SIP Phone Integration

[Madovsky]

depend the transcoding, from 10 to 20


  ----- Original Message ----- 
  From: James Mbuthia 
  To: FreeSWITCH Users Help 
  Sent: Monday, December 06, 2010 10:23 AM
  Subject: Re: [Freeswitch-users] SIP Phone Integration


  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 
      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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