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<DIV><FONT size=2>depend the transcoding, from 10 to 20</FONT></DIV>
<DIV><FONT size=2></FONT> </DIV>
<DIV> </DIV>
<BLOCKQUOTE
style="BORDER-LEFT: #000000 2px solid; PADDING-LEFT: 5px; PADDING-RIGHT: 0px; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px">
<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 Help</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Monday, December 06, 2010 10:23
AM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [Freeswitch-users] SIP Phone
Integration</DIV>
<DIV><BR></DIV>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
style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex"
class=gmail_quote>
<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>
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