hi Kristian,<br><br>what a relief for me to browse your stuff. the RFC docs are daunting for me. this is a nice companion to the RFCs. now i can understand the SIP messages in FS much better. keep it up. i'll give u feedback whatever i see needs correction. and ... THANKS A LOT!<br>
<br>-nandy<br><br><div class="gmail_quote">On Fri, Jun 15, 2012 at 4:06 AM, Kristian Kielhofner <span dir="ltr"><<a href="mailto:kris@kriskinc.com" target="_blank">kris@kriskinc.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
FreeSWITCHers,<br>
<br>
As some of you may have noticed I frequently comment on SIP-related<br>
FreeSWITCH posts. Long story short I decided to finally put some of<br>
these thoughts and experiences down in writing. What I ended up with<br>
is a 21 page (and counting) document describing various SIP header<br>
fields, SDP information, RTP issues, DTMF issues, NAT traversal<br>
technologies, interop headaches, etc.<br>
<br>
I wrote a little intro and linked to the document if any of you<br>
would like to check it out (link at the end of post):<br>
<br>
<a href="http://blog.krisk.org/2012/06/everything-you-wish-you-didnt-need-to.html" target="_blank">http://blog.krisk.org/2012/06/everything-you-wish-you-didnt-need-to.html</a><br>
<br>
Of course I'd appreciate any feedback you may have. What do you<br>
disagree with? Find a typo? Have burning SIP/RTP/SDP/T.38/codec<br>
questions you'd like to see me address? Should I get into more<br>
implementation (FreeSWITCH) specific issues? Let me know.<br>
<br>
Thanks!<br>
<br>
--<br>
Kristian Kielhofner<br>
<br>
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</blockquote></div><br>