SIP session termination doesn't work properly

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SIP session termination doesn't work properly

Simon Witte
Hi all,


I'm having trouble with the proper termination of SIP sessions going
through my proxy server (opensips 1.5.1)
There are 2 scenarios so far:

1) Session between two hardware audiocodec (Mayah Centauri)
    The BYE from one client isn't answered by a "200 - OK" but a "404 -
Not here". First one client tries 12 times, then the other one.

    screenshots from wireshark:
http://www.vivid-vision.de/Files/bye_404_no1.jpg
                                                 
http://www.vivid-vision.de/Files/bye_404_no2.jpg

2) Session between two laptops using a sip-software
    the BYE seems to hop from one client to the other until it times out.
   
    screenshot from wireshark:
http://www.vivid-vision.de/Files/bye_timeout.jpg


Can somebody please tell if there's something wrong with the routing
logic or has a hint to what the problem might be?
I'm not so sure yet, if it's an opensips-problem, but I want to rule it
out first.

Thank you all in advance.


Best regards,
Simon


CONFIG!


#
# $Id: opensips.cfg 5503 2009-03-22 16:22:32Z bogdan_iancu $
#
# OpenSIPS basic configuration script
#     by Anca Vamanu <[hidden email]>
#
# Please refer to the Core CookBook at:
#      http://www.opensips.org/index.php?n=Resources.DocsCookbooks
# for a explanation of possible statements, functions and parameters.
#


####### Global Parameters #########


debug=3
log_stderror=no
log_facility=LOG_LOCAL0

fork=yes
children=4

port=5060

#listen=udp:192.168.1.60:5060
#listen=tcp:192.168.1.60:5060


####### Modules Section ########

mpath="/usr/local/lib/opensips/modules/"

loadmodule "db_mysql.so"
loadmodule "signaling.so"
loadmodule "sl.so"
loadmodule "tm.so"
loadmodule "rr.so"
loadmodule "maxfwd.so"
loadmodule "usrloc.so"
loadmodule "registrar.so"
loadmodule "textops.so"
loadmodule "mi_fifo.so"
loadmodule "uri_db.so"
loadmodule "uri.so"
loadmodule "xlog.so"
loadmodule "acc.so"
loadmodule "auth.so"
loadmodule "auth_db.so"


/* uncomment next line for aliases support
   NOTE: a DB (like db_mysql) module must be also loaded */
loadmodule "alias_db.so"

/* uncomment next line for multi-domain support
   NOTE: a DB (like db_mysql) module must be also loaded
   NOTE: be sure and enable multi-domain support in all used modules
         (see "multi-module params" section ) */

loadmodule "domain.so"

/* uncomment the next two lines for presence server support
   NOTE: a DB (like db_mysql) module must be also loaded */

#loadmodule "presence.so"
#loadmodule "presence_xml.so"




# ----------------- setting module-specific parameters ---------------


# ----- mi_fifo params -----

modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")


# ----- rr params -----

modparam("rr", "enable_full_lr", 1)
modparam("rr", "append_fromtag", 0)
modparam("rr", "enable_double_rr", 0)
modparam("rr", "add_username", 0)


# ----- registrar params -----

modparam("registrar", "method_filtering", 1)
modparam("registrar", "append_branches", 0)
modparam("registrar", "max_contacts", 10)
modparam("registrar", "default_expires", 900)
modparam("registrar", "min_expires", 60)
modparam("registrar", "max_expires", 0)
modparam("registrar", "case_sensitive", 0)
modparam("registrar", "default_q", 0)
modparam("registrar", "received_param", "received")


# ----- usrloc params -----

modparam("usrloc", "desc_time_order", 1)
modparam("usrloc", "user_column", "username")
modparam("usrloc", "domain_column", "domain")
modparam("usrloc", "contact_column", "contact")
modparam("usrloc", "expires_column", "expires")
modparam("usrloc", "q_column", "q")
modparam("usrloc", "callid_column", "callid")
modparam("usrloc", "cseq_column", "cseq")
modparam("usrloc", "methods_column", "methods")
modparam("usrloc", "flags_column", "flags")
modparam("usrloc", "user_agent_column", "user_agent")
modparam("usrloc", "received_column", "received")
modparam("usrloc", "socket_column", "socket")
modparam("usrloc", "use_domain", 1)
modparam("usrloc", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips")
modparam("usrloc", "db_mode", 2)
modparam("usrloc", "matching_mode", 0)
modparam("usrloc", "timer_interval", 15)
modparam("usrloc", "cseq_delay", 5)



# ----- uri_db params -----

modparam("uri_db", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips")
modparam("uri_db", "use_domain", 0)
modparam("uri_db", "user_column", "username")
modparam("uri_db", "domain_column", "domain")


# ----- db_mysql params -----

modparam("db_mysql", "ping_interval", 300)
modparam("db_mysql", "auto_reconnect", 1)


# ----- maxfwd params -----

modparam("maxfwd", "max_limit", 256)


# ----- acc params -----

/* what sepcial events should be accounted ? */
modparam("acc", "early_media", 1)
modparam("acc", "report_ack", 1)
modparam("acc", "report_cancels", 1)
/* by default ww do not adjust the direct of the sequential requests.
   if you enable this parameter, be sure the enable "append_fromtag"
   in "rr" module */
modparam("acc", "detect_direction", 0)
/* account triggers (flags) */
modparam("acc", "failed_transaction_flag", 3)
modparam("acc", "log_flag", 1)
modparam("acc", "log_missed_flag", 2)
/* uncomment the following lines to enable DB accounting also */
modparam("acc", "db_flag", 1)
modparam("acc", "db_missed_flag", 2)




# ----- auth_db params -----

modparam("auth_db", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips")
modparam("auth_db", "user_column", "username")
modparam("auth_db", "domain_column", "domain")
modparam("auth_db", "calculate_ha1", 1)
modparam("auth_db", "password_column", "password")
modparam("auth_db", "password_column_2", "ha1b")
modparam("auth_db", "use_domain", 1)
modparam("auth_db", "load_credentials", "rpid")


# ----- alias_db params -----

/* uncomment the following lines if you want to enable the DB based
   aliases */
#modparam("alias_db", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips")


# ----- domain params -----

#modparam("domain", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips")
#modparam("domain", "db_mode", 1)   # Use caching


# ----- multi-module params -----

/* uncomment the following line if you want to enable multi-domain support
   in the modules (dafault off) */
#modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1)


# ----- presence params -----

#modparam("presence|presence_xml", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips")
#modparam("presence_xml", "force_active", 1)
#modparam("presence", "server_address", "sip:192.168.1.2:5060")




####### Routing Logic ########


# main request routing logic

route{

    # --1-- request validation

    if (!mf_process_maxfwd_header("10")) {
        sl_send_reply("483","Too Many Hops");
        exit;
    }

    if (msg:len >= 2048) {
        sl_send_reply("513", "Message too big");
        exit;
    }
   

    # --2-- routing preprocessing

    if(is_method("OPTIONS")) {
        sl_send_reply("200", "OK");
        exit;
    }

     
    if (has_totag()) {
        # sequential request withing a dialog should
        # take the path determined by record-routing
        if (loose_route()) {
            if (is_method("BYE")) {
                setflag(1); # do accounting ...
                setflag(3); # ... even if the transaction fails
            } else if (is_method("INVITE")) {
                # even if in most of the cases is useless, do RR for
                # re-INVITEs also, as some buggy clients do change route set
                # during the dialog.
                record_route();
            }
            # route it out to whatever destination was set by loose_route()
            # in $du (destination URI).
            route(1);
        } else {
            /* uncomment the following lines if you want to enable
presence */
            ##if (is_method("SUBSCRIBE") && $rd ==
"your.server.ip.address") {
            ##    # in-dialog subscribe requests
            ##    route(2);
            ##    exit;
            ##}
            if ( is_method("ACK") ) {
                if ( t_check_trans() ) {
                    # non loose-route, but stateful ACK; must be an ACK
after
                    # a 487 or e.g. 404 from upstream server
                    t_relay();
                    exit;
                } else {
                    # ACK without matching transaction ->
                    # ignore and discard
                    exit;
                }
            }
            sl_send_reply("404","Not here");
        }
        exit;
    }

    #initial requests

    # CANCEL processing
    if (is_method("CANCEL"))
    {
        if (t_check_trans())
            t_relay();
        exit;
    }

    t_check_trans();

    # authenticate if from local subscriber (uncomment to enable auth)
    # authenticate all initial non-REGISTER request that pretend to be
    # generated by local subscriber (domain from FROM URI is local)
    if (!(method=="REGISTER") && from_uri==myself) /*no multidomain
version*/
    ##if (!(method=="REGISTER") && is_from_local())  /*multidomain version*/
    {
        if (!proxy_authorize("192.168.1.60", "subscriber")) {
            proxy_challenge("192.168.1.60", "0");
            exit;
        }
        if (!check_from()) {
            sl_send_reply("403","Forbidden auth ID");
            exit;
        }
   
        consume_credentials();
        # caller authenticated
   
    }

    # preloaded route checking
    if (loose_route()) {
        xlog("L_ERR",
        "Attempt to route with preloaded Route's [$fu/$tu/$ru/$ci]");
        if (!is_method("ACK"))
            sl_send_reply("403","Preload Route denied");
        exit;
    }

    # record routing
    if (!is_method("REGISTER"))
        record_route();

    # account only INVITEs
    if (is_method("INVITE")) {
        setflag(1); # do accounting
    }
    if (!uri==myself)
    ## replace with following line if multi-domain support is used
    ##if (!is_uri_host_local())
    {
        append_hf("P-hint: outbound\r\n");
        # if you have some interdomain connections via TLS
        ##if($rd=="tls_domain1.net") {
        ##    t_relay("tls:domain1.net");
        ##    exit;
        ##} else if($rd=="tls_domain2.net") {
        ##    t_relay("tls:domain2.net");
        ##    exit;
        ##}
        route(1);
    }

    # requests for my domain

    ## uncomment this if you want to enable presence server
    ##   and comment the next 'if' block
    ##   NOTE: uncomment also the definition of route[2] from  below
    ##if( is_method("PUBLISH|SUBSCRIBE"))
    ##        route(2);

    if (is_method("PUBLISH"))
    {
        sl_send_reply("503", "Service Unavailable");
        exit;
    }
   

    if (is_method("REGISTER"))
    {
        # authenticate the REGISTER requests (uncomment to enable auth)
        if (!www_authorize("192.168.1.60", "subscriber"))
        {
            www_challenge("192.168.1.60", "0");
            exit;
        }
       
        if (!check_to())
        {
            sl_send_reply("403","Forbidden auth ID");
            exit;
        }

        if (!save("location"))
            sl_reply_error();

        exit;

    }
   


    if ($rU==NULL) {
        # request with no Username in RURI
        sl_send_reply("484","Address Incomplete");
        exit;
    }

    # apply DB based aliases (uncomment to enable)
    ##alias_db_lookup("dbaliases");

    if (!lookup("location")) {
        switch ($retcode) {
            case -1:
            case -3:
                t_newtran();
                t_reply("404", "Not Found");
                exit;
            case -2:
                sl_send_reply("405", "Method Not Allowed");
                exit;
        }
    }

    # when routing via usrloc, log the missed calls also
    setflag(2);

    route(1);
}


route[1] {
    # for INVITEs enable some additional helper routes
    if (is_method("INVITE")) {
        t_on_branch("2");
        t_on_reply("2");
        t_on_failure("1");
    }

    if (!t_relay()) {
        sl_reply_error();
    };
    exit;
}


# Presence route
/* uncomment the whole following route for enabling presence
   NOTE: do not forget to enable the call of this route from the main
     route */
##route[2]
##{
##    if (!t_newtran())
##    {
##        sl_reply_error();
##        exit;
##    };
##
##    if(is_method("PUBLISH"))
##    {
##        handle_publish();
##        t_release();
##    }
##    else
##    if( is_method("SUBSCRIBE"))
##    {
##        handle_subscribe();
##        t_release();
##    }
##
##    exit;
##}


branch_route[2] {
    xlog("new branch at $ru\n");
}


onreply_route[2] {
    xlog("incoming reply\n");
}


failure_route[1] {
    if (t_was_cancelled()) {
        exit;
    }

    # uncomment the following lines if you want to block client
    # redirect based on 3xx replies.
    ##if (t_check_status("3[0-9][0-9]")) {
    ##t_reply("404","Not found");
    ##    exit;
    ##}

    # uncomment the following lines if you want to redirect the failed
    # calls to a different new destination
    ##if (t_check_status("486|408")) {
    ##    sethostport("192.168.2.100:5060");
    ##    # do not set the missed call flag again
    ##    t_relay();
    ##}
}



--
___________________________________
Simon Witte


phone: +49 211 - 545 888 02
mobil: +49 172 - 286 20 76

mail: [hidden email]

Engelbertstr. 11 | 40233 Düsseldorf




__________ Hinweis von ESET NOD32 Antivirus, Signaturdatenbank-Version 4081 (20090517) __________

E-Mail wurde geprüft mit ESET NOD32 Antivirus.

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Re: SIP session termination doesn't work properly

Bogdan-Andrei Iancu
Hi Simon,

could you post the SIP capture with full message (use ngrep on the
machine) - otherwise it it rather impossible to figure out what is going
on there.

Regards,
Bogdan

Simon Witte wrote:

> Hi all,
>
>
> I'm having trouble with the proper termination of SIP sessions going
> through my proxy server (opensips 1.5.1)
> There are 2 scenarios so far:
>
> 1) Session between two hardware audiocodec (Mayah Centauri)
>     The BYE from one client isn't answered by a "200 - OK" but a "404 -
> Not here". First one client tries 12 times, then the other one.
>
>     screenshots from wireshark:
> http://www.vivid-vision.de/Files/bye_404_no1.jpg
>                                                  
> http://www.vivid-vision.de/Files/bye_404_no2.jpg
>
> 2) Session between two laptops using a sip-software
>     the BYE seems to hop from one client to the other until it times out.
>    
>     screenshot from wireshark:
> http://www.vivid-vision.de/Files/bye_timeout.jpg
>
>
> Can somebody please tell if there's something wrong with the routing
> logic or has a hint to what the problem might be?
> I'm not so sure yet, if it's an opensips-problem, but I want to rule it
> out first.
>
> Thank you all in advance.
>
>
> Best regards,
> Simon
>
>
>  


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Re: SIP session termination doesn't work properly

Simon Witte
Hi Bogdan,

I just captured the two sessions with ngrep (attached to this mail). Suprisingly, the dialog between the two laptops is all good with ngrep.
But when I use the two mayah centauri hardware codecs, there is still the problem with the "404 - not here". Which brings me back to my assumption that it is not a opensips problem. Can you confirm that?

Thanks & best regards,
Simon



-------- Original-Nachricht --------
> Datum: Mon, 18 May 2009 17:45:54 +0300
> Von: Bogdan-Andrei Iancu <[hidden email]>
> An: [hidden email]
> CC: [hidden email]
> Betreff: Re: [OpenSIPS-Users] SIP session termination doesn\'t work properly

> Hi Simon,
>
> could you post the SIP capture with full message (use ngrep on the
> machine) - otherwise it it rather impossible to figure out what is going
> on there.
>
> Regards,
> Bogdan
>
> Simon Witte wrote:
> > Hi all,
> >
> >
> > I'm having trouble with the proper termination of SIP sessions going
> > through my proxy server (opensips 1.5.1)
> > There are 2 scenarios so far:
> >
> > 1) Session between two hardware audiocodec (Mayah Centauri)
> >     The BYE from one client isn't answered by a "200 - OK" but a "404 -
> > Not here". First one client tries 12 times, then the other one.
> >
> >     screenshots from wireshark:
> > http://www.vivid-vision.de/Files/bye_404_no1.jpg
> >                                                  
> > http://www.vivid-vision.de/Files/bye_404_no2.jpg
> >
> > 2) Session between two laptops using a sip-software
> >     the BYE seems to hop from one client to the other until it times
> out.
> >    
> >     screenshot from wireshark:
> > http://www.vivid-vision.de/Files/bye_timeout.jpg
> >
> >
> > Can somebody please tell if there's something wrong with the routing
> > logic or has a hint to what the problem might be?
> > I'm not so sure yet, if it's an opensips-problem, but I want to rule it
> > out first.
> >
> > Thank you all in advance.
> >
> >
> > Best regards,
> > Simon
> >
> >
> >  
--
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two_codecs_404_error.txt (41K) Download Attachment
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Re: SIP session termination doesn't work properly

Bogdan-Andrei Iancu
Hi Simon,

The User-Agent: MAYAH Communications V2.0.0.44 (callee party) is bogus
as it does not mirror the RR headers from the received INVITE.

It receives:

U 192.168.1.60:5060 -> 192.168.1.70:5060
INVITE sip:Centauri@192.168.1.70:5060 SIP/2.0.
Record-Route: <sip:192.168.1.60;lr=on;ftag=ZoSgtn2eqKF>.
....


but the 200 OK

U 192.168.1.70:5060 -> 192.168.1.60:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.1.60;branch=z9hG4bK6e58.04604ae3.0.
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bKp65iCSWas2K;received=192.168.1.50;rport=5060.
Max-Forwards: 70.
To: <sip:mayah2@192.168.1.60>;tag=SetXW2miM8y.
From: <sip:mayah1@192.168.1.60>;tag=ZoSgtn2eqKF.
Call-ID: ZoSgtn2e1242662605@192.168.1.50.
CSeq: 2 INVITE.
Contact: sip:Centauri@192.168.1.70:5060.
User-Agent: MAYAH Communications V2.0.0.44.
Content-Length: 289.
Content-Type: application/sdp.
....

has not RR header mirrored (as per RFC3261) back to the caller - because
of this, the BYE cannot be properly routed.

Regards,
Bogdan

Simon Witte wrote:

> Hi Bogdan,
>
> I just captured the two sessions with ngrep (attached to this mail). Suprisingly, the dialog between the two laptops is all good with ngrep.
> But when I use the two mayah centauri hardware codecs, there is still the problem with the "404 - not here". Which brings me back to my assumption that it is not a opensips problem. Can you confirm that?
>
> Thanks & best regards,
> Simon
>
>
>
> -------- Original-Nachricht --------
>  
>> Datum: Mon, 18 May 2009 17:45:54 +0300
>> Von: Bogdan-Andrei Iancu <[hidden email]>
>> An: [hidden email]
>> CC: [hidden email]
>> Betreff: Re: [OpenSIPS-Users] SIP session termination doesn\'t work properly
>>    
>
>  
>> Hi Simon,
>>
>> could you post the SIP capture with full message (use ngrep on the
>> machine) - otherwise it it rather impossible to figure out what is going
>> on there.
>>
>> Regards,
>> Bogdan
>>
>> Simon Witte wrote:
>>    
>>> Hi all,
>>>
>>>
>>> I'm having trouble with the proper termination of SIP sessions going
>>> through my proxy server (opensips 1.5.1)
>>> There are 2 scenarios so far:
>>>
>>> 1) Session between two hardware audiocodec (Mayah Centauri)
>>>     The BYE from one client isn't answered by a "200 - OK" but a "404 -
>>> Not here". First one client tries 12 times, then the other one.
>>>
>>>     screenshots from wireshark:
>>> http://www.vivid-vision.de/Files/bye_404_no1.jpg
>>>                                                  
>>> http://www.vivid-vision.de/Files/bye_404_no2.jpg
>>>
>>> 2) Session between two laptops using a sip-software
>>>     the BYE seems to hop from one client to the other until it times
>>>      
>> out.
>>    
>>>    
>>>     screenshot from wireshark:
>>> http://www.vivid-vision.de/Files/bye_timeout.jpg
>>>
>>>
>>> Can somebody please tell if there's something wrong with the routing
>>> logic or has a hint to what the problem might be?
>>> I'm not so sure yet, if it's an opensips-problem, but I want to rule it
>>> out first.
>>>
>>> Thank you all in advance.
>>>
>>>
>>> Best regards,
>>> Simon
>>>
>>>
>>>  
>>>      
>
>  


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