[Freeswitch-users] SIP Phone Integration

[Madovsky]

or use acme company,
but very expensive
  ----- Original Message ----- 
  From: David Ponzone 
  To: FreeSWITCH Users Help 
  Sent: Monday, December 06, 2010 12:56 PM
  Subject: Re: [Freeswitch-users] SIP Phone Integration


  If you choose another codec, you could use a HW-based transcoding solution like the Sangoma D100/500, which goes from 30 to 2000 concurrent calls.
  According their website, they don't support Speex.


  David Ponzone  Direction Technique
  email: david.ponzone at ipeva.fr
  tel:      01 74 03 18 97
  gsm:   06 66 98 76 34


  Service Client IPeva
  tel:      0811 46 26 26
  www.ipeva.fr  -   www.ipeva-studio.com


  Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.







  Le 06/12/2010 à 16:23, James Mbuthia a écrit :


    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










------------------------------------------------------------------------



        _______________________________________________
        FreeSWITCH-users mailing list
        FreeSWITCH-users at lists.freeswitch.org
        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        http://www.freeswitch.org




      _______________________________________________
      FreeSWITCH-users mailing list
      FreeSWITCH-users at lists.freeswitch.org
      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      http://www.freeswitch.org



    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org





------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20101206/7b7b013c/attachment-0001.html