[Kamailio-Users] switch to TCP when UDP message is bigger than MTU

Daniel-Constantin Mierla miconda at gmail.com
Tue Apr 21 14:05:43 CEST 2009


Hello,

On 04/20/2009 08:34 PM, Klaus Darilion wrote:
> Hi Sandro!
>
> If you just need it for testing then you could use the approaches 
> mentioned in the other emails.
>
> But for a production system you usually do not want this feature. 
> Practically this paragraph from RFC 3261 is non-sense as it brakes 
> communication with SIP clients.
>
> Residential customers SIP clients are mostly behind NAT and often do not 
> even support TCP. Thus, automatic switching to TCP will cause problems 
> as you can not reach the clients anymore.
>
> Usually administrators look how to turn this feature off instead of 
> turning it on.
>   
is it supposed to happen even if UA binds location with UDP contact? I 
thought is used manly for inter-proxy/scb communication.

Cheers,
Daniel

> regards
> Klaus
>
> Sandro Bordacchini schrieb:
>   
>> Hi all.
>>
>> I am doing some test about tcp/udp transport protocols in SIP and IP 
>> fragmentation.
>>
>> I would like to test this behaviour, as stated in par. 18.1.1 of rfc3261:
>>
>>    If a request is within 200 bytes of the path MTU, or if it is larger
>>    than 1300 bytes and the path MTU is unknown, the request MUST be sent
>>    using an RFC 2914 [43] congestion controlled transport protocol, such
>>    as TCP. If this causes a change in the transport protocol from the
>>    one indicated in the top Via, the value in the top Via MUST be
>>    changed.  This prevents fragmentation of messages over UDP and
>>    provides congestion control for larger messages.  However,
>>    implementations MUST be able to handle messages up to the maximum
>>    datagram packet size.
>>
>> To do this, I am running a OpenSer 1.3.2 with a "default" configuration 
>> (and very simple: no accounting, no auth, no DB) and I have two SIP 
>> phones registered on that server (for example, ext. 100 and ext. 101).
>>
>> When 100 calls 101, the INVITE reaches the OpenSer that appends some 
>> "dummy" headers (I have added some append_hf in the route), just to let 
>> the message be bigger than the MTU.
>> The INVITE is routed towards 101 with UDP protocol and I have IP 
>> fragmentation.
>>
>> Do you have some hint to get the OpenSer work as described in the RFC 
>> (i.e. switch automatically to TCP)?
>> Or is this only possible with UACs and not with proxies?
>>
>> TIA
>>
>>     
>
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>   

-- 
Daniel-Constantin Mierla
http://www.asipto.com/





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