[OpenSIPS-Users] sip message enters on bucle

Bogdan-Andrei Iancu bogdan at opensips.org
Thu Apr 5 11:55:07 CEST 2012


Hi Jorge,

No, it is not a bug - what is going on on your side is perfectly normal. 
The root problem is that the TCP connection (from behind a NAT) which 
was used when the call is established, this conn is down at BYE time and 
cannot be re-open by opensips....

Best regards,
Bogdan

On 04/04/2012 06:48 PM, Jorge Ortea wrote:
> Hi Bogdan,
>
> Ok, now we known that is happening. But, is it logic? or is it a bug 
> in 1.6.4.2 version?
>
> Curiously this does not happen with UDP signaling.
>
> Thanks.
> Regards.
>
> ------------------------------------------------------------------------
> Date: Wed, 4 Apr 2012 18:19:04 +0300
> From: bogdan at opensips.org
> To: darham at hotmail.com
> CC: users at lists.opensips.org
> Subject: Re: [OpenSIPS-Users] sip message enters on bucle
>
> Jorge,
>
> the message is not looping, it is retransmitting - it is something 
> different. OpenSIPS tries to open a new TCP conn to the destination 
> (as there is no existing one), but it fails in timeout as you cannot 
> open a TCP conn somewhere behind a NAT.
>
> Regards,
> Bogdan
>
> On 04/04/2012 06:06 PM, Jorge Ortea wrote:
>
>
>     Hi Bogdan,
>
>     Is correct, Z.Z.Z.Z:5062 is a public adress behind a NAT. I have
>     found that opensips haven't this tcp connection, now this account
>     has changed the public adress.
>
>     But the sip messages keeps in the loop. It's like if Opensips is
>     looking for a tcp connection that it hasn't.... ?¿
>
>     Thanks.
>     Regards.
>
>
>     ------------------------------------------------------------------------
>     Date: Wed, 4 Apr 2012 17:38:31 +0300
>     From: bogdan at opensips.org <mailto:bogdan at opensips.org>
>     To: darham at hotmail.com <mailto:darham at hotmail.com>
>     CC: users at lists.opensips.org <mailto:users at lists.opensips.org>
>     Subject: Re: [OpenSIPS-Users] sip message enters on bucle
>
>     Hi Jorge,
>
>     So opensips tries to send the BYE to Z.Z.Z.Z:5062 via TCP (guess
>     based on Route hdrs), but nobody is listening on TCP - is this
>     address pointing behind a NAT ? why is not accepting a new TCP
>     connection.
>
>     On the other side, what you can do is to reduce the timeout on TCP
>     connection, so opensips will react sooner:
>     http://www.opensips.org/Resources/DocsCoreFcn18#toc78
>
>     Regards,
>     Bogdan
>
>     On 04/04/2012 05:16 PM, Jorge Ortea wrote:
>
>
>         Hi Bogdan,
>
>         Exactly, is ready, OpenSIPS try to reach to destination but
>         now the account 2105 haven't the location:  Z.Z.Z.Z:5062
>
>         In fact, when OpenSIPS try to reach to there, it write in
>         log:     (this account uses TLS signaling)
>
>         Apr  4 10:14:17 alpha02 /usr/local/sbin/opensips[29503]:
>         :::::: BYE - from 911111111 to O2105 - Callid:
>         5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152
>         <mailto:5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152> - Source:
>         X.X.X.152
>         Apr  4 10:14:18 alpha02 /usr/local/sbin/opensips[29525]:
>         :::::: BYE - from 911111111 to O2105 - Callid:
>         5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152
>         <mailto:5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152> - Source:
>         X.X.X.152
>         Apr  4 10:14:19 alpha02 /usr/local/sbin/opensips[29497]:
>         :::::: BYE - from 911111111 to O2105 - Callid:
>         5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152
>         <mailto:5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152> - Source:
>         X.X.X.152
>         Apr  4 10:14:21 alpha02 /usr/local/sbin/opensips[29487]:
>         :::::: BYE - from 911111111 to O2105 - Callid:
>         5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152
>         <mailto:5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152> - Source:
>         X.X.X.152
>         Apr  4 10:14:25 alpha02 /usr/local/sbin/opensips[29511]:
>         :::::: BYE - from 911111111 to O2105 - Callid:
>         5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152
>         <mailto:5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152> - Source:
>         X.X.X.152
>         Apr  4 10:14:27 alpha02 /usr/local/sbin/opensips[29503]:
>         ERROR:core:tcp_blocking_connect: timeout 10 s elapsed from 10 s
>         Apr  4 10:14:27 alpha02 /usr/local/sbin/opensips[29503]:
>         ERROR:core:tcpconn_connect: tcp_blocking_connect failed
>         Apr  4 10:14:27 alpha02 /usr/local/sbin/opensips[29503]:
>         ERROR:core:tcp_send: connect failed
>         Apr  4 10:14:27 alpha02 /usr/local/sbin/opensips[29503]:
>         ERROR:tm:msg_send: tcp_send failed
>         Apr  4 10:14:27 alpha02 /usr/local/sbin/opensips[29503]:
>         ERROR:tm:t_forward_nonack: sending request failed
>
>         Thus, how can i detect and avoid this ??
>
>         Thanks.
>         Regards.
>
>
>         ------------------------------------------------------------------------
>         Date: Wed, 4 Apr 2012 14:56:16 +0300
>         From: bogdan at opensips.org <mailto:bogdan at opensips.org>
>         To: users at lists.opensips.org <mailto:users at lists.opensips.org>
>         CC: darham at hotmail.com <mailto:darham at hotmail.com>
>         Subject: Re: [OpenSIPS-Users] sip message enters on bucle
>
>         Hi Jorge,
>
>         It looks like Asterisk generates the BYEs and retransmits it
>         because there is no reply coming back from opensips. Normally
>         the BYE is end 2 end replied (so the other end device should
>         generate the reply for BYE).
>         But looking at the 477 reply you get from OpenSIPS, I suspect
>         that OpenSIPS was trying to forward the BYE request (maybe via
>         TCP), got blocked and failed at the end - this failure
>         resulted in the 477 reply.
>
>         Check the opensips logs to see error when processing the BYE.
>
>         Regards,
>         Bogdan
>
>         On 04/04/2012 11:42 AM, Jorge Ortea wrote:
>
>             Hi,
>
>             I have the follow VoIP platform;  OpenSIPS 1.6.4.2-tls +
>             Mediaproxy 2.0 + a pair of Asterisks 1.4 (behind SER)
>
>             It works fine but sometimes a sip message enters on a
>             loop. Asterisk sends 5 sip messages at every turn
>
>
>             My logs in OpenSIPS:
>
>             Apr  4 10:14:17 alpha02 /usr/local/sbin/opensips[29503]:
>             :::::: BYE - from 911111111 to O2105 - Callid:
>             5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152
>             <mailto:5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152> -
>             Source: X.X.X.152
>             Apr  4 10:14:18 alpha02 /usr/local/sbin/opensips[29525]:
>             :::::: BYE - from 911111111 to O2105 - Callid:
>             5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152
>             <mailto:5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152> -
>             Source: X.X.X.152
>             Apr  4 10:14:19 alpha02 /usr/local/sbin/opensips[29497]:
>             :::::: BYE - from 911111111 to O2105 - Callid:
>             5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152
>             <mailto:5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152> -
>             Source: X.X.X.152
>             Apr  4 10:14:21 alpha02 /usr/local/sbin/opensips[29487]:
>             :::::: BYE - from 911111111 to O2105 - Callid:
>             5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152
>             <mailto:5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152> -
>             Source: X.X.X.152
>             Apr  4 10:14:25 alpha02 /usr/local/sbin/opensips[29511]:
>             :::::: BYE - from 911111111 to O2105 - Callid:
>             5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152
>             <mailto:5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152> -
>             Source: X.X.X.152
>
>
>
>             Sip messages in Asterisk *CLI> 'sip debug':
>
>             set_destination: Parsing
>             <sip:X.X.X.150;r2=on;lr=on;did=e25.7093e044> for
>             address/port to send to
>             set_destination: set destination to X.X.X.150, port 5060
>             Reliably Transmitting (no NAT) to X.X.X.150:5060:
>             BYE sip:2105 at Z.Z.Z.Z:5062;transport=tls SIP/2.0
>             Via: SIP/2.0/UDP X.X.X.152:5060;branch=z9hG4bK59044fff;rport
>             Route:
>             <sip:X.X.X.150;r2=on;lr=on;did=e25.7093e044>,<sip:Y.Y.Y.150:5061;transport=tls;r2=on;lr=on;did=e25.7093e044>
>             From: "911111111" <sip:911111111 at X.X.X.152>;tag=as167eb28e
>             To: <sip:O2105 at X.X.X.150>;tag=bcd482cd12b8a21i0
>             Call-ID: 5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152
>             <mailto:5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152>
>             CSeq: 2874 BYE
>             User-Agent: Asterisk PBX
>             Max-Forwards: 70
>             X-Asterisk-HangupCause: Normal Clearing
>             X-Asterisk-HangupCauseCode: 16
>             Content-Length: 0
>
>
>             ---
>             Scheduling destruction of SIP dialog
>             '5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152
>             <mailto:5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152>' in
>             32000 ms (Method: REFER)
>             Retransmitting #1 (no NAT) to X.X.X.150:5060:
>             BYE sip:2105 at Z.Z.Z.Z:5062;transport=tls SIP/2.0
>             Via: SIP/2.0/UDP X.X.X.152:5060;branch=z9hG4bK59044fff;rport
>             Route:
>             <sip:X.X.X.150;r2=on;lr=on;did=e25.7093e044>,<sip:Y.Y.Y.150:5061;transport=tls;r2=on;lr=on;did=e25.7093e044>
>             From: "911111111" <sip:911111111 at X.X.X.152>;tag=as167eb28e
>             To: <sip:O2105 at X.X.X.150>;tag=bcd482cd12b8a21i0
>             Call-ID: 5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152
>             <mailto:5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152>
>             CSeq: 2874 BYE
>             User-Agent: Asterisk PBX
>             Max-Forwards: 70
>             X-Asterisk-HangupCause: Normal Clearing
>             X-Asterisk-HangupCauseCode: 16
>             Content-Length: 0
>
>
>             ---
>             Retransmitting #2 (no NAT) to X.X.X.150:5060:
>             BYE sip:2105 at Z.Z.Z.Z:5062;transport=tls SIP/2.0
>             Via: SIP/2.0/UDP X.X.X.152:5060;branch=z9hG4bK59044fff;rport
>             Route:
>             <sip:X.X.X.150;r2=on;lr=on;did=e25.7093e044>,<sip:Y.Y.Y.150:5061;transport=tls;r2=on;lr=on;did=e25.7093e044>
>             From: "911111111" <sip:911111111 at X.X.X.152>;tag=as167eb28e
>             To: <sip:O2105 at X.X.X.150>;tag=bcd482cd12b8a21i0
>             Call-ID: 5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152
>             <mailto:5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152>
>             CSeq: 2874 BYE
>             User-Agent: Asterisk PBX
>             Max-Forwards: 70
>             X-Asterisk-HangupCause: Normal Clearing
>             X-Asterisk-HangupCauseCode: 16
>             Content-Length: 0
>
>
>             ---
>             Retransmitting #3 (no NAT) to X.X.X.150:5060:
>             BYE sip:2105 at Z.Z.Z.Z:5062;transport=tls SIP/2.0
>             Via: SIP/2.0/UDP X.X.X.152:5060;branch=z9hG4bK59044fff;rport
>             Route:
>             <sip:X.X.X.150;r2=on;lr=on;did=e25.7093e044>,<sip:Y.Y.Y.150:5061;transport=tls;r2=on;lr=on;did=e25.7093e044>
>             From: "911111111" <sip:911111111 at X.X.X.152>;tag=as167eb28e
>             To: <sip:O2105 at X.X.X.150>;tag=bcd482cd12b8a21i0
>             Call-ID: 5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152
>             <mailto:5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152>
>             CSeq: 2874 BYE
>             User-Agent: Asterisk PBX
>             Max-Forwards: 70
>             X-Asterisk-HangupCause: Normal Clearing
>             X-Asterisk-HangupCauseCode: 16
>             Content-Length: 0
>
>
>             ---
>             Retransmitting #4 (no NAT) to X.X.X.150:5060:
>             BYE sip:2105 at Z.Z.Z.Z:5062;transport=tls SIP/2.0
>             Via: SIP/2.0/UDP X.X.X.152:5060;branch=z9hG4bK59044fff;rport
>             Route:
>             <sip:X.X.X.150;r2=on;lr=on;did=e25.7093e044>,<sip:Y.Y.Y.150:5061;transport=tls;r2=on;lr=on;did=e25.7093e044>
>             From: "911111111" <sip:911111111 at X.X.X.152>;tag=as167eb28e
>             To: <sip:O2105 at X.X.X.150>;tag=bcd482cd12b8a21i0
>             Call-ID: 5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152
>             <mailto:5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152>
>             CSeq: 2874 BYE
>             User-Agent: Asterisk PBX
>             Max-Forwards: 70
>             X-Asterisk-HangupCause: Normal Clearing
>             X-Asterisk-HangupCauseCode: 16
>             Content-Length: 0
>
>
>             ---
>
>             <--- SIP read from X.X.X.150:5060 --->
>             SIP/2.0 477 Send failed (477/TM)
>             Via: SIP/2.0/UDP
>             X.X.X.152:5060;branch=z9hG4bK59044fff;rport=5060
>             From: "911111111" <sip:911111111 at X.X.X.152>;tag=as167eb28e
>             To: <sip:O2105 at X.X.X.150>;tag=bcd482cd12b8a21i0
>             Call-ID: 5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152
>             <mailto:5b62cc795e6be4ea3fa9a26e543e3622 at X.X.X.152>
>             CSeq: 2874 BYE
>             Server: OpenSIPS (1.6.4-2-tls (i386/linux))
>             Content-Length: 0
>
>
>             <------------->
>             --- (8 headers 0 lines) ---
>             SIP Response message for INCOMING dialog BYE arrived
>                 -- Incoming call: Got SIP response 477 "Send failed
>             (477/TM)" back from X.X.X.150
>
>
>
>             At the end, i have restart the asterisk to solve it. How
>             can I avoid it ?
>
>
>             Thanks.
>             Regards.
>

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

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.opensips.org/pipermail/users/attachments/20120405/a8375e79/attachment-0001.htm>


More information about the Users mailing list