[SR-Users] How to disable retransmits via TCP connection? etc.

[Daniel-Constantin Mierla]

Have you grabbed the sip trace on client side to see what it is
receiving? Are the clients reporting errors?

If you have a snom phone, you can easily see the received sip packets
via web interface. Perhaps the desktop phones will have also some logs
printing what is happening that can be accessed easily.

Eventually you can try to run a kamailio locally, near the client, using
it as an intermediate proxy between the phone and the main sip server.

The timestamps I checked in previous traces were not following the sip
retransmissions intervals (0.5sec, 1sec, 2sec, ...), a clear indication
that it is not kamailio transaction layer doing retransmissions.

As I said before, ngrep is not a source to trust when dealing with large
packets. Also, it can happen that it prints the same packet twice.

Daniel

On 23/06/15 17:32, Andrey Utkin wrote:
> 2015-06-19 12:45 GMT+03:00 Daniel-Constantin Mierla <miconda at gmail.com>:
>> Also, ngrep is not always good at capturing big packets, so just seeing
>> partial sip packets is ok as long as the receiving party doesn't
>> complain of broken/incomplete sip packet.
> Thank you Daniel for answering.
> We still struggle from this weird issue. Now I can reproduce it in my
> location with recent Kamailio from git, with all SIP over TCP clients
> - Linphone, Sipdroid, Jitsi (desktop Java-based app).
> Here are ngrep logs from both server and client sides, and server
> syslog containing Kamailio output (filtered by substring "tcp"
> case-insensitively).
> https://gist.github.com/krieger-od/55427f2b3923b910bacb
> https://gist.github.com/krieger-od/c9fe6ea4bb64fac82cda
> https://gist.github.com/krieger-od/96ef40ca15ef2407b5f4
>
> Here you can see that only a "tail" of INVITE message gets transmitted
> to client side, which is weird.
> This is Amazon server; it have MTU 9001 by default, but setting it to
> 900 or 1100 haven't made any difference. Also I have reproduced this
> issue on DigitalOcean VPS, so this mustn't be Amazon-specific issue.
> Because I have tried different SIP useragents supporting TCP, I'm
> afraid this can be considered Kamailio issue (honestly, I still don't
> quite believe as I percept Kamailio as robust and stable software).
> Any review and comment helps.
>

-- 
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.com