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<DIV><FONT size=2>You can transcode with FS, </FONT></DIV>
<DIV><FONT size=2>I did already a web phone (boophone) with FS but ;keep in
mind</FONT></DIV>
<DIV><FONT size=2>that transcode=CPU</FONT></DIV>
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<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="FONT: 10pt arial; BACKGROUND: #e4e4e4; font-color: black"><B>From:</B>
<A title=jmmbuthia@gmail.com href="mailto:jmmbuthia@gmail.com">James
Mbuthia</A> </DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=freeswitch-users@lists.freeswitch.org
href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Monday, December 06, 2010 9:49
AM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> [Freeswitch-users] SIP Phone
Integration</DIV>
<DIV><BR></DIV>
<DIV class=gmail_quote>
<DIV class=gmail_quote>
<DIV class=gmail_quote>
<DIV>
<DIV class=im>
<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>
<P>
<HR>
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