Yes indeed, we mainly use TCP as our SIP INVITE conveys a XML document with list of callees (>UDP MTU) and our presence notfications are also big messages.<div>Pascal<br><br><div class="gmail_quote">On Wed, Apr 14, 2010 at 11:51 AM, Daniel-Constantin Mierla <span dir="ltr"><<a href="mailto:miconda@gmail.com">miconda@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
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Hi Pascal,<div class="im"><br>
<br>
On 4/14/10 11:42 AM, Pascal Maugeri wrote:
<blockquote type="cite">Hi Daniel
<div><br>
</div>
<div>Just to let you know I followed your advice and we deployed
kamailio 3.0.1. </div>
<div>We are still doing several tests, but I can say already we don't
have anymroe errors with tcp connections ...</div>
</blockquote></div>
ok, thanks for feedback. Indeed, 3.0+ is far more improved than what
was in openser/kamailio 1.0 to 1.5 in respect to tcp, therefore anyone
facing heavy tcp needs should consider this 3.0+.<br>
<br>
Cheers,<br>
Daniel<div><div></div><div class="h5"><br>
<br>
<blockquote type="cite">
<div><br>
</div>
<div>Cheers</div>
<div>Pascal</div>
<div><br>
<br>
<div class="gmail_quote">On Fri, Mar 12, 2010 at 6:04 PM,
Daniel-Constantin Mierla <span dir="ltr"><<a href="mailto:miconda@gmail.com" target="_blank">miconda@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left:1px solid rgb(204, 204, 204);margin:0pt 0pt 0pt 0.8ex;padding-left:1ex">Hello,
<div>
<div><br>
<br>
On 03/11/2010 05:58 PM, Iņaki Baz Castillo wrote:<br>
<blockquote class="gmail_quote" style="border-left:1px solid rgb(204, 204, 204);margin:0pt 0pt 0pt 0.8ex;padding-left:1ex">
2010/3/11 Pascal Maugeri<<a href="mailto:pascal.maugeri@gmail.com" target="_blank">pascal.maugeri@gmail.com</a>>:<br>
<br>
<blockquote class="gmail_quote" style="border-left:1px solid rgb(204, 204, 204);margin:0pt 0pt 0pt 0.8ex;padding-left:1ex">
<blockquote class="gmail_quote" style="border-left:1px solid rgb(204, 204, 204);margin:0pt 0pt 0pt 0.8ex;padding-left:1ex">Does
such NOTIFY go to a TCP registered user? Of course if there is<br>
not an existing TCP connection between Kamailio and the final natted<br>
user then it's not possible to send such NOTIFY.<br>
<br>
<br>
</blockquote>
Do you mean that the user is sending "transport=tcp" in his Contact
header ?<br>
<br>
</blockquote>
This must be present in the initial SUBSCRIBE. However if the client<br>
is behind NAT and uses TCP it's required some way to mantain the<br>
keepalive in the router, if not a future NOTIFY could not arrive. A<br>
common approach is the client sending some TCP data through the<br>
existing connection (i.e.<CRLF><CRLF> as defined in
defat-oubound,<br>
now RFC XXXX).<br>
<br>
</blockquote>
</div>
</div>
I have seen clients sending registration over UDP requiring to be
contacted via TCP.<br>
<br>
To be sure it registers via TCP check the configuration of the phone
and watch the sip traffic with ngrep (or ethereal) to see the transport
layer protocol.<br>
<br>
Connecting from server to a client behind nat is possible only if you
have port forwarding on your nat box to phone IP address. Therefore, if
the phone connects via tcp it must keep the connection open. If for
some reason it closes, it must re-open it. Otherwise it becomes
unreachable.<br>
<br>
In the server side there are lot of tcp options to tune the behavior
and optimize. I do suggest using version 3.0 for a much improved TCP
architecture and implementation (including asynchronous tcp -- in case
you deal with lot of tcp connections, then this saves you).<br>
<a href="http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:3.0.x#tcp_parameters" target="_blank">http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:3.0.x#tcp_parameters</a><br>
<br>
Worth to mention as well that you can change the value of tcp
parameters at runtime without need to restart (e.g., connecting
timeout, send timeout, etc) using sercmd.<br>
<br>
Cheers,<br>
Daniel<br>
<font color="#888888"><br>
-- <br>
Daniel-Constantin Mierla<br>
Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010<br>
* <a href="http://www.asipto.com/index.php/sip-router-masterclass/" target="_blank">http://www.asipto.com/index.php/sip-router-masterclass/</a><br>
<br>
</font></blockquote>
</div>
<br>
</div>
</blockquote>
<br>
</div></div><font color="#888888"><div>-- <br>
Daniel-Constantin Mierla
* <a href="http://www.asipto.com/" target="_blank">http://www.asipto.com/</a> * <a href="http://twitter.com/miconda" target="_blank">http://twitter.com/miconda</a>
* <a href="http://www.linkedin.com/in/danielconstantinmierla" target="_blank">http://www.linkedin.com/in/danielconstantinmierla</a>
</div>
</font></div>
</blockquote></div><br></div>