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

Daniel-Constantin Mierla miconda at gmail.com
Fri Jun 19 11:45:09 CEST 2015


Hello,

On 19/06/15 11:29, Andrey Utkin wrote:
> Hi!
> We meet such an issue with some kinds of client internet connections
> (e.g. reproduces always on local wifi, but doesn't repr. on 3g inet).
> We don't know what are exact network characteristics, but what happens
> from perspective of traffic on SIP server is the following:
>
> 1. the SIP client registers successfully to Kamailio (which is recent
> git master HEAD) with TCP transport for SIP;
> 2. another client calls this user;
> 3. Kamailio relays INVITE, it gets transmitted fully and ACK-ed on TCP level;
> 4. then, within 0.1-0.2 second, Kamailio fires first retransmit of
> that INVITE, which doesn't get fully transmitted (it is shown
> trunkated in sniffer output, no TCP ACK replies for it);
> 5. Kamailio tries to retransmit INVITE again, with the same result as in #5.
> 6. The caller gets SIP/2.0 408 Request Timeout.
>
> A piece of traffic from "ngrep -t -e -d any -W byline port 5060":
> https://gist.githubusercontent.com/krieger-od/219f9975e5efb980ff5b/raw/096a639bf459b5f7c8a2a83eba8424f0dd3d22ea/gistfile1.txt
>
> The called side app is based on mobile Linphone app.
> Switching to UDP is not an option (in some networks SIP messages get
> delivered trunkated which breaks calls), we will check how it works
> with TLS transport a bit later when there's technical possibility.
>
> I have two questions:
> 1. How would I completely disable retransmissions to TCP connections?
> 2. Any ideas what can be the reason for this issue? Retransmissions by
> themselves?
>
there should be no retransmission on TCP done by Kamailio.

I a look at time stamps, it is different than standard SIP
retransmission intervals. Have you changed them in tm parameters?

As you say it is over wireless, might be the network driver doing some
re-tranmissions.

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.

Cheers,
Daniel

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




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