[SR-Users] t_relay dying ?
Jean Cérien
cerien.jean at gmail.com
Tue Jan 2 14:52:20 CET 2018
Thanks
The INVITE is received from a VOIP provider, on the kamailio box, and
forwarded to the asterisk box. For all my voip provider, it works fine, but
just for one, the ACK is not passed onto the asterisk box, but a trace
shows that the ACK is sent by kamailio to kamaiio - I have attached below
the SIP packets for the original invite and the ACK that gets incorrectly
forwarded,
J.
2017/12/26 15:21:47.414524 VOIP-IP:5060 -> KAMAILIOIP:5060
INVITE sip:CALLED-NUM at KAMAILIOIP SIP/2.0
Via: SIP/2.0/UDP VOIP-IP:5060;branch=z9hG4bKh5r5mp20007mno86oit0.1
P-Asserted-Identity: <sip:CALLING-NUM at orange-multimedia.fr>
To: <sip:CALLED-NUM at KAMAILIOIP:5060>
From: <sip:CALLING-NUM at orange-multimedia.fr>;tag=SDf1f0501-
496399c4-0016-096a-0000-0000
Call-ID: SDf1f0501-70980730711ea436a8ed241985dddc7d-v300g00040
CSeq: 1 INVITE
Max-Forwards: 65
Contact: <sip:CALLING-NUM at VOIP-IP:5060;transport=udp>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS
Supported: uui,timer
Accept: application/sdp, application/isup, application/xml
P-Access-Network-Info: GSTN;operator-specific-GI="
639720000";network-provided
Content-Type: application/sdp
Content-Length: 221
Session-Expires: 1800; refresher=uac
Min-SE: 90
*ACK Received from provider*
2017/12/26 15:21:47.473423 VOIP-IP:5060 -> KAMAILIOIP:5060
ACK sip:CALLED-NUM at KAMAILIOIP:5060 SIP/2.0
Via: SIP/2.0/UDP VOIP-IP:5060;branch=z9hG4bKmdfnm110182k1pbvl180.1
To: <sip:KAMAILIOIP:5060>;tag=as3eec8631
From: <sip:CALLING-NUM at orange-multimedia.fr>;tag=SDf1f0501-
496399c4-0016-096a-0000-0000
Call-ID: SDf1f0501-70980730711ea436a8ed241985dddc7d-v300g00040
CSeq: 1 ACK
Max-Forwards: 66
Content-Length: 0
Route: <sip:KAMAILIOIP;lr=on;ftag=SDf1f0501-496399c4-0016-096a-
0000-0000;did=eea.aa52>
*ACK Forwarded*
2017/12/26 15:21:47.473898* KAMAILIOIP:5060 -> KAMAILIOIP:5060*
ACK sip:KAMAILIOIP;lr=on;ftag=SDf1f0501-496399c4-0016-096a-0000-0000;did=eea.aa52
SIP/2.0
Via: SIP/2.0/UDP KAMAILIOIP;branch=z9hG4bK52bb.
2e7f3f0b7c8c60b92bfa2d945cc9829d.0
Via: SIP/2.0/UDP VOIP-IP:5060;branch=z9hG4bKmdfnm110182k1pbvl180.1
To: <sip:KAMAILIOIP:5060>;tag=as3eec8631
From: <sip:CALLING-NUM at orange-multimedia.fr>;tag=SDf1f0501-
496399c4-0016-096a-0000-0000
Call-ID: SDf1f0501-70980730711ea436a8ed241985dddc7d-v300g00040
CSeq: 1 ACK
Max-Forwards: 65
Content-Length: 0
On Tue, Jan 2, 2018 at 9:19 AM, Wilkins, Steve <swwilkins at mitre.org> wrote:
> Hello,
>
>
>
> Have you done a trace tcpdump/Wireshark? What does the traffic look
> like? Are Asterisk and Kamailio on different Servers?
>
>
>
> -Steve
>
>
>
> *From:* sr-users [mailto:sr-users-bounces at lists.kamailio.org] *On Behalf
> Of *Jean Cérien
> *Sent:* Tuesday, January 2, 2018 8:04 AM
> *To:* Daniel-Constantin Mierla <miconda at gmail.com>
> *Cc:* Kamailio (SER) - Users Mailing List <sr-users at lists.kamailio.org>
> *Subject:* Re: [SR-Users] t_relay dying ?
>
>
>
>
>
> Hello
>
>
>
> Any suggestion as to why the ACK is not forwarded to the Asterisk box and
> stays on the kamailio server ?
>
>
>
> REgards
>
> J.
>
>
>
> On Tue, Dec 26, 2017 at 11:49 AM, Jean Cérien <cerien.jean at gmail.com>
> wrote:
>
>
>
> Daniel
>
>
>
> Many thanks for your help in this holidays season.
>
>
>
> The ACK is now going out and executions follows, which is good, however,
> the ack is sent to the Kamailio box itself and not actually forwarded to
> the asterisk (I've attached at the end of this message the initial invite,
> received ACK and "forwarded" ACK).
>
>
>
> Both packet go through very classic config, starting with dlg_manage(),
> then if (has_totag()), if (loose_route()), which leads to route(RELAY),
> which does the actual call to t_relay.
>
>
>
> This sequence works fine with other providers, I cant see what is wrong !
>
>
>
> Regards
>
> J
>
>
>
>
>
> *Initial invite received from provider*
>
>
>
> 2017/12/26 15:21:47.414524 VOIP-IP:5060 -> KAMAILIOIP:5060
>
> INVITE sip:CALLED-NUM at KAMAILIOIP SIP/2.0
>
> Via: SIP/2.0/UDP VOIP-IP:5060;branch=z9hG4bKh5r5mp20007mno86oit0.1
>
> P-Asserted-Identity: <sip:CALLING-NUM at orange-multimedia.fr>
>
> To: <sip:CALLED-NUM at KAMAILIOIP:5060>
>
> From: <sip:CALLING-NUM at orange-multimedia.fr>;tag=SDf1f0501-
> 496399c4-0016-096a-0000-0000
>
> Call-ID: SDf1f0501-70980730711ea436a8ed241985dddc7d-v300g00040
>
> CSeq: 1 INVITE
>
> Max-Forwards: 65
>
> Contact: <sip:CALLING-NUM at VOIP-IP:5060;transport=udp>
>
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS
>
> Supported: uui,timer
>
> Accept: application/sdp, application/isup, application/xml
>
> P-Access-Network-Info: GSTN;operator-specific-GI="
> 639720000";network-provided
>
> Content-Type: application/sdp
>
> Content-Length: 221
>
> Session-Expires: 1800; refresher=uac
>
> Min-SE: 90
>
>
>
>
>
> *ACK Received from provider*
>
>
>
> 2017/12/26 15:21:47.473423 VOIP-IP:5060 -> KAMAILIOIP:5060
>
> ACK sip:CALLED-NUM at KAMAILIOIP:5060 SIP/2.0
>
> Via: SIP/2.0/UDP VOIP-IP:5060;branch=z9hG4bKmdfnm110182k1pbvl180.1
>
> To: <sip:KAMAILIOIP:5060>;tag=as3eec8631
>
> From: <sip:CALLING-NUM at orange-multimedia.fr>;tag=SDf1f0501-
> 496399c4-0016-096a-0000-0000
>
> Call-ID: SDf1f0501-70980730711ea436a8ed241985dddc7d-v300g00040
>
> CSeq: 1 ACK
>
> Max-Forwards: 66
>
> Content-Length: 0
>
> Route: <sip:KAMAILIOIP;lr=on;ftag=SDf1f0501-496399c4-0016-096a-
> 0000-0000;did=eea.aa52>
>
>
>
>
>
> *ACK Forwarded*
>
>
>
> 2017/12/26 15:21:47.473898* KAMAILIOIP:5060 -> KAMAILIOIP:5060*
>
> ACK sip:KAMAILIOIP;lr=on;ftag=SDf1f0501-496399c4-0016-096a-
> 0000-0000;did=eea.aa52 SIP/2.0
>
> Via: SIP/2.0/UDP KAMAILIOIP;branch=z9hG4bK52bb.
> 2e7f3f0b7c8c60b92bfa2d945cc9829d.0
>
> Via: SIP/2.0/UDP VOIP-IP:5060;branch=z9hG4bKmdfnm110182k1pbvl180.1
>
> To: <sip:KAMAILIOIP:5060>;tag=as3eec8631
>
> From: <sip:CALLING-NUM at orange-multimedia.fr>;tag=SDf1f0501-
> 496399c4-0016-096a-0000-0000
>
> Call-ID: SDf1f0501-70980730711ea436a8ed241985dddc7d-v300g00040
>
> CSeq: 1 ACK
>
> Max-Forwards: 65
>
> Content-Length: 0
>
>
>
>
>
>
>
>
>
> On Sat, Dec 23, 2017 at 8:12 AM, Daniel-Constantin Mierla <
> miconda at gmail.com> wrote:
>
> I just tested and works fine. Try with master branch and default config
> file.
>
> To test I modified a bit the default kamailio.cfg and I have in
> route[RELAY]:
>
> xlog("=======* request $rm before relay\n");
> if (!t_relay()) {
> xlog("=======x request $rm not relayed\n");
> sl_reply_error();
> }
> xlog("=======> request $rm relayed\n");
>
> I get the ACK routed and in logs I get:
>
> {1 2 ACK 1473723014 at 127.0.0.1} 2(20129) ERROR: <script>: =======*
> request ACK before relay
> {1 2 ACK 1473723014 at 127.0.0.1} 2(20129) ERROR: <script>: =======>
> request ACK relayed
>
> Cheers,
> Daniel
>
>
>
> On 22.12.17 21:00, Jean Cérien wrote:
>
>
>
> Many thanks for your help.
>
> I've applied the patch, recompiled and I see no difference unfortunately.
> The ACK is not forwarded, and execution after t_relay stops.
>
>
>
> I've set the debug level to 4 and captured the traces - I believe the
> section dealing with the ACK is there: https://pastebin.com/TG03YQWC
>
>
>
> KR
>
> J.
>
>
>
> On Fri, Dec 22, 2017 at 11:59 AM, Daniel-Constantin Mierla <
> miconda at gmail.com> wrote:
>
> Hello,
>
> can you give a try with latest master branch or by using the patch from
> next link?
>
> - https://github.com/kamailio/kamailio/commit/
> 52111974b4571e0562e8e731df80f48dbc504915
>
> Apparently the t_realy() stopped script execution (by returning 0) when
> e2e ACK forwarding was successful. The patch should change that to return
> true.
>
> If all works fine while testing, then I will backport.
>
> Cheers,
> Daniel
>
>
>
>
>
> On 22.12.17 14:05, Jean Cérien wrote:
>
>
>
> Hello
>
> I've tried mhomed = 1 with no luck - not so sure what you mean by default
> route, I've added listen = udp:myadsress:5060
>
> I also have auto_aliases=no and alias=myaddress
>
>
>
> However, what bugs me is that execution seems to stop INSIDE t_relay and
> the info message after the t_relay is not displayed.
>
>
>
> I hate to say this, but could this be a bug ?
>
>
>
> Rgds
>
> J.
>
>
>
>
>
> On Fri, Dec 22, 2017 at 1:52 AM, Sergey Safarov <s.safarov at gmail.com>
> wrote:
>
> Check that your installation have one NIC with only one default route on
> host.
>
> If not check that "mhomed=1" is enabled.
>
>
>
> Sergey
>
>
>
> пт, 22 дек. 2017 г. в 0:02, Jean Cérien <cerien.jean at gmail.com>:
>
>
>
> Hello
>
> I am using kamailio 5.0.2, on a debian 9 system.
>
>
>
> Everything was running fine, until one of our voip provider changed his
> switch. Our kamailio is relaying between several voip providers and several
> asterisk (only the signalisation, no rtp).
>
>
>
> When we get an invite from this new switch, we select an asterisk and
> relay it correctly to this box. However, the OK is relayed back. But when
> the voip providers sends an ACK for this OK, the t_relay function does not
> return at all, it just dies with no action and no error.
>
>
>
> Here is the snippet from route(relay)
>
>
>
> xlog("L_INFO","route(relay) @@ $rm - Source: $si:$sp, fu:$fu,
> tu:$tu\n" );
>
> $var(restrelay)=t_relay();
>
> xlog("L_INFO","route(relay) @@ $rm - t_relay result:
> $var(restrelay)" );
>
> if (!$var(restrelay)) {
>
>
>
> When processing the initial invite, I do get both INFO messages. When the
> ACK is processed, I only get one INFO message, and no ACK is relayed - so
> it seems execution dies in the t_relay
>
>
>
> What could be wrong ???
>
>
>
> J.
>
>
>
> Details of the ACK:
>
>
>
> ACK sip:dialednumber at kamailioIP:5060 SIP/2.0
>
> Via: SIP/2.0/UDP VOIPPROVIDERIP:5060;branch=z9hG4bKjq16fi00d85uee181ic0.1
>
> To: <sip:kamailioIP:5060>;tag=as7f10ed48
>
> From: <sip:sourcenumber at orange-multimedia.fr>;tag=SD8u3ob01-
> 7dd8efde-0002-00d5-0000-0000
>
> Call-ID: SD8u3ob01-688d41f5e2164ceeafe76f40b82b3f97-v300g00030
>
> CSeq: 1 ACK
>
> Max-Forwards: 66
>
> Content-Length: 0
>
> Route: <sip:kamailioIP;lr=on;ftag=SD8u3ob01-7dd8efde-0002-00d5-
> 0000-0000;did=a0b.49d>
>
>
>
>
>
>
>
> (should you have enough width; the diagram will look ok)
>
>
>
> VOIP PROVIDER KAMAILIO
> ASTERISK
>
> 21:38:54.685149 │ ──────────────────────────> │
> │
>
> ▒ +0.000451 │ 100 trying -- your call is │
> │
>
> ▒ 21:38:54.685600 │ <────────────────────────── │
> │
>
> ▒ +0.000084 │ │ INVITE (SDP)
> │
>
> ▒ 21:38:54.685684 │ │
> ──────────────────────────> │
>
> ▒ +0.000831 │ │ 100 Trying
> │
>
> ▒ 21:38:54.686515 │ │
> <────────────────────────── │
>
> ▒ +0.000471 │ │ 200 OK (SDP)
> │
>
> ▒ 21:38:54.686986 │ │
> <────────────────────────── │
>
> ▒ +0.000394 │ 200 OK (SDP) │
> │
>
> ▒ 21:38:54.687380 │ <────────────────────────── │
> │
>
> ▒ +0.038694 │ ACK │
> │
>
> ▒ 21:38:54.726074 │ ──────────────────────────> │
> │
>
> ▒ +0.060155 │ │ 200 OK (SDP)
> │
>
> ▒ 21:38:54.786229 │ │
> <<<──────────────────────── │
>
> ▒ +0.000138 │ 200 OK (SDP) │
> │
>
> ▒ 21:38:54.786367 │ <<<──────────────────────── │
> │
>
> ▒ +0.005721 │ ACK │
> │
>
>
>
>
>
> _______________________________________________
> Kamailio (SER) - Users Mailing List
> sr-users at lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
>
> _______________________________________________
> Kamailio (SER) - Users Mailing List
> sr-users at lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
>
>
> _______________________________________________
>
> Kamailio (SER) - Users Mailing List
>
> sr-users at lists.kamailio.org
>
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
> --
>
> Daniel-Constantin Mierla
>
> www.twitter.com/miconda -- www.linkedin.com/in/miconda
>
> Kamailio Advanced Training - www.asipto.com
>
> Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com
>
>
>
>
>
> --
>
> Daniel-Constantin Mierla
>
> www.twitter.com/miconda -- www.linkedin.com/in/miconda
>
> Kamailio Advanced Training - www.asipto.com
>
> Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com
>
>
>
>
>
> _______________________________________________
> Kamailio (SER) - Users Mailing List
> sr-users at lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
>
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