Thanks, about how many concurrent calls are you able to when transcoding with FS?<br><br><div class="gmail_quote">On Mon, Dec 6, 2010 at 5:17 PM, Madovsky <span dir="ltr"><<a href="mailto:infos@madovsky.org">infos@madovsky.org</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div bgcolor="#ffffff">
<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>
<blockquote style="border-left:#000000 2px solid;padding-left:5px;padding-right:0px;margin-left:5px;margin-right:0px"><div><div></div><div class="h5">
<div style="font:10pt arial">----- Original Message ----- </div>
<div style="font:10pt arial;background:#e4e4e4"><b>From:</b>
<a title="jmmbuthia@gmail.com" href="mailto:jmmbuthia@gmail.com" target="_blank">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" target="_blank">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>
<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>
</div></div><p>
</p><hr>
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