[OpenSIPS-Users] uac_auth

Bogdan-Andrei Iancu bogdan at opensips.org
Mon Nov 4 12:25:30 CET 2013


Hi Rik,

The truth is in the middle. The second invite from opensips (the one
with credentials) must not be considered a retransmission - it has a
totally different VIA branch -> different transaction.
Also, OpenSIPS should increase the CSeq when answering to the challenge,
but not able to do so as OpenSIPS is mainly a SIP proxy, not a b2bua.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 11/04/2013 12:22 PM, Rik Broers wrote:
>
> Hello Bogdan,
>
>  
>
> Yes I'm very sure that the proper credentials are used ;)
>
>  
>
> I'm going to try and calculate the response according to the RFC.
>
>  
>
> One thing I found is that asterisk seems to ignore my second invite
> with Authorization because of retransmit?
>
> It seems that I should increase my CSEQ on second invite.. How can I
> do this neatly?
>
>  
>
> [Nov  4 11:08:25] DEBUG[22804]: chan_sip.c:22448 handle_incoming: ****
> Received INVITE (5) - Command in SIP INVITE
>
> [Nov  4 11:08:25] DEBUG[22804]: chan_sip.c:22467 handle_incoming:
> Ignoring SIP message because of retransmit (INVITE Seqno 12481, ours
> 12481) Ignoring this INVITE request
>
>  
>
>  
>
> Met vriendelijke groet,
>
>  
>
> *Rik Broers*
>
> */Voice Engineer/**//*
>
>  
>
>  
>
> *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
> *Sent:* vrijdag 1 november 2013 12:34
> *To:* Rik Broers
> *Cc:* OpenSIPS users mailling list
> *Subject:* Re: [OpenSIPS-Users] uac_auth
>
>  
>
> Hello Rik,
>
> It may be silly , but are you sure you filled in the proper
> credentials (realm, auth user and password) ??
>
> Also, based on how the response for digest is computed, you can double
> check the OpenSIPS auth response (calculating the HA and md5 sums as
> per RFC 2617).
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
>
> On 11/01/2013 01:09 PM, Rik Broers wrote:
>
>     Yes, thats correct. Opensips sends out an invite with
>     Authorization header as response on the 401 unauthorized.
>
>     This authorization header contains the correct Nonce.
>
>     Instead of being authorized I receive another 401 unauthorized
>     which opensips replies again with new nonce and so on until max
>     branches is reached.
>
>      
>
>     Met vriendelijke groet,
>
>     Regards,
>
>      
>
>     *Rik Broers*
>
>     */Voice Engineer/*
>
>      
>
>      
>
>     *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
>     *Sent:* vrijdag 1 november 2013 11:49
>     *To:* OpenSIPS users mailling list
>     *Cc:* Rik Broers
>     *Subject:* Re: [OpenSIPS-Users] uac_auth
>
>      
>
>     Hello Rik,
>
>     So OpenSIPS generates a new INVITE with credentials (as a result
>     of the uac_auth() ), but this is also rejected ?
>
>     Regards,
>
>
>     Bogdan-Andrei Iancu
>
>     OpenSIPS Founder and Developer
>
>     http://www.opensips-solutions.com
>
>
>     On 10/31/2013 11:46 AM, Rik Broers wrote:
>
>         Hi,
>
>          
>
>         I'm trying to use the uac_auth() function to add Authorization
>         to my invite after I received a 401 Unauthorized.
>
>         I call the function in the failure route and according to
>         Debug the authorization header is inserted. I also see this in
>         a trace.
>
>         Unfortunately I haven't been able to authorize successfully,
>         double checked everything and also tried with phones to ensure
>         the credentials are correct and my asterisk is working.
>
>         I'm filling the credentials with a modparam not with AVP.
>
>          
>
>         In DBG I see this: DBG:uac_auth:build_authorization_hdr: hdr
>         is <Authorization: Digest username="**", realm="**",
>         nonce="31d5b0d9", uri="***;user=phone",
>         response="ea344343187f27c668be8fdc3acf8c5a",
>         algorithm=MD5#015#012>
>
>         So it seems to match correctly.
>
>          
>
>         I'm authenticating against Asterisk. And my failure route
>         looks like this:
>
>         failure_route[FailPBX]{
>
>                 xlog("Im in failpbx route");
>
>                 uac_auth();
>
>                 t_on_failure("FailPBX");
>
>                 t_relay();
>
>         }
>
>          
>
>         What happens is the following
>
>         -> Invite
>
>         <- 100 Giving a try
>
>         <- 401 Unauthorized (Unique nonce 1)
>
>         -> ACK
>
>         -> invite with authorization header (unique Nonce 1)
>
>         <- 100 Giving a try
>
>         <- 401 Unauthorized (Unique nonce 2)
>
>         -> invite with authorization header (unique Nonce 2)
>
>         ..... and so on until ERROR:tm:add_uac: maximum number of
>         branches exceeded.
>
>          
>
>          
>
>         Only thing left for me now is to verify that the Digest
>         calculated is correct. *How can I do this?* What functions
>         should I use on linux..
>
>         Below my authorization challenge.
>
>         imap://bogdan@opensips.org:993/fetch%3EUID%3E.INBOX%3E191220?header=quotebody&part=1.2&filename=image005.png
>
>          
>
>         Or are there any other things I'm missing?
>
>         Im using NOTICE:core:main: version: opensips 1.10.0-notls
>         (x86_64/linux)
>
>          
>
>          
>
>         Met vriendelijke groet,
>
>         Regards,
>
>          
>
>         *Rik Broers*
>
>         */Voice Engineer/*
>
>          
>
>          
>
>          
>
>         _______________________________________________
>
>         Users mailing list
>
>         Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>
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>
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