[OpenSIPS-Users] What is protocol/port mismatch?

[Bogdan-Andrei Iancu]

Hi Brian,

Regarding the crash you mentioned - do you have any backtraces ?

About some of your doubts:
1) t_relay() is not forcing any proto by itself: it preserves the 
inbound proto if the RURI (or socket) is not saying otherwise.
2) turning off the double RR may brake some things as opensips will not 
be able to properly route between the original inbound and outbound 
interfaces....only if you do it manually from the script.

To me, the solution looks like a dirty hack, but if makes Asterisk 
happy....what can I say :)

Regards,
Bogdan

opensipslist at encambio.com wrote:
> Hello list,
>
> The problem was that the voicemail server was respecting the host
> and port of the routeset which OpenSIPS was rewriting to acommodate
> the transport change from TLS to TCP. The voicemail server was
> configured to send over TLS to the outbound proxy however, so
> traffic was arriving at OpenSIPS on the port 5060 using TLS.
>
> It's clear now why the warnings were appearing in the logs, but
> even more it seems that OpenSIPS sometimes crashes when sending
> TLS traffic to a TCP port which it is expecting clear text from.
>
> To know about the solution to this problem, read below.
>
> An ven., janv 29, 2010, opensipslist at encambio.com schrieb:
>   
>> An ven., janv 29, 2010, Bogdan-Andrei Iancu schrieb:
>>     
>>> opensipslist at encambio.com wrote:
>>>       
>>>> ...and I see this in the log:
>>>>
>>>>   <warning> opensips[2717]: WARNING:core:get_send_socket: protocol/port mismatch
>>>>
>>>> some hundreds of times per day (about once every 20 minutes
>>>> per registered UAC.)
>>>>
>>>> I have this in the route script:
>>>>
>>>>   listen = tls:name.host.tld:5061
>>>>   [...]
>>>>   t_relay("name.host.tld:5080");
>>>>
>>>>         
> First, although all UAs were exchanging SIP traffic with OpenSIPS
> over TLS, t_relay was relaying the messages over UDP. I assume that
> t_relay is either hardcoded to use UDP or OpenSIPS was routing the
> messages to itself internally over UDP (to recursively resolve
> aliases.) In this case maybe t_relay implicitly uses the transport
> over which the message was last received (even internally.)
>
> In any case, the call to t_relay should look like this if relaying
> over the TCP transport is desired:
>
>   t_relay("tcp:name.host.tld:5080");
>
> The problem then is that the rr module will add two Record-Route
> headers, expecting a symetrical fashion of SIP traffic. I turned
> this off with modparam("rr", "enable_double_rr", 0) because the
> voicemail server (Asterisk) is very tricky (maybe even buggy) to
> configure to use TLS and outbound proxies.
>
> At this point I saw that traffic was arriving at OpenSIPS port 5061
> over TLS as expected, and as indicated by the routeset. The problem
> is solved.
>
>   
>> Is the basic idea of this warning message that OpenSIPS
>> exchanges a SIP message with a UA over a certain transport
>> (TLS in this case) and port number (5061 in this case), but
>> a t_relay in the route script forwards the message over a
>> different transport or to a different port number?
>>
>>     
> The basic idea of this warning message is that OpenSIPS expects
> a certain type (TLS or not TLS) of traffic at a certain TCP port
> as configured by the 'listen' directives. If a UAC sends TLS
> traffic to a port only configured to listen to plain text traffic
> then this warning will appear. What happens to the mismatched
> traffic is not clear, but I assume that it is ignored by OpenSIPS.
>
>   
>> If so, what is being compared and how is it compared?
>>
>>     
> The TCP port on which OpenSIPS listens to and the type of traffic,
> either TLS encrypted or plain text.
>
> Regards,
> Brian
>
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>
>   


-- 
Bogdan-Andrei Iancu
www.voice-system.ro