14 Apr
2010
14 Apr
'10
1:08 p.m.
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.
Pascal
On Wed, Apr 14, 2010 at 11:51 AM, Daniel-Constantin Mierla <
miconda(a)gmail.com> wrote:
Hi Pascal,
On 4/14/10 11:42 AM, Pascal Maugeri wrote:
Hi Daniel
Just to let you know I followed your advice and we deployed kamailio
3.0.1.
We are still doing several tests, but I can say already we don't have
anymroe errors with tcp connections ...
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+.
Cheers,
Daniel
Cheers
Pascal
On Fri, Mar 12, 2010 at 6:04 PM, Daniel-Constantin Mierla <
miconda(a)gmail.com> wrote:
Hello,
On 03/11/2010 05:58 PM, Iñaki Baz Castillo wrote:
--
Daniel-Constantin Mierla * http://www.asipto.com/ *
http://twitter.com/miconda *
http://www.linkedin.com/in/danielconstantinmierla
2010/3/11 Pascal
Maugeri<pascal.maugeri(a)gmail.com>om>:
I have seen clients sending registration over UDP requiring to be
contacted via TCP.
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.
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.
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).
http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:3.0.x#tcp_parameters
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.
Cheers,
Daniel
--
Daniel-Constantin Mierla
Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010
* http://www.asipto.com/index.php/sip-router-masterclass/
Does such NOTIFY go to a TCP registered user? Of
course if there is
> not an existing TCP connection between Kamailio and the final natted
> user then it's not possible to send such NOTIFY.
>
>
>
Do you mean that the user is sending "transport=tcp" in his Contact
header ?
This must be present in the initial SUBSCRIBE. However if the client
is behind NAT and uses TCP it's required some way to mantain the
keepalive in the router, if not a future NOTIFY could not arrive. A
common approach is the client sending some TCP data through the
existing connection (i.e.<CRLF><CRLF> as defined in defat-oubound,
now RFC XXXX).