already been done, check out red5phone. i have it working w/ FS and red5 server<br><br><div class="gmail_quote">On Mon, Dec 6, 2010 at 9:49 AM, James Mbuthia <span dir="ltr"><<a href="mailto:jmmbuthia@gmail.com">jmmbuthia@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"><div class="gmail_quote"><div class="gmail_quote"><div class="gmail_quote"><div><div><div>Hi guys,<br>
<br>
Am new to Freeswitch and am looking for info that can help me develop a web-based SIP Phone<br>
<br>
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. </div><div><br></div></div><div>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.</div>
<div><br></div><div>regards,</div><div>
James Mbuthia<br>
</div></div></div><br>
</div><br>
</div><br>
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