GRUU contact not found

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GRUU contact not found

Robert Dyck

My understanding is that GRUU processing in opensips is automatic, provided it is not disabled. No further configuration or scripting is required. Is that correct.

 

A GRUU capable UA rergisters and receives public and temporary GR identities. The UA establishes a dialog with another UA. The callee ends the call. The caller does not recive the BYE.

 

Caller :

Request-Line: INVITE sip:7@192.168.1.2 SIP/2.0

Contact URI: sip:4@192.168.1.2:5060;gr=urn:uuid:35dfa98a-2feb-482a-bde7-7568a86348b1

 

Callee:

Status-Line: SIP/2.0 200 OK

 

Caller:

Request-Line: ACK sip:7@192.168.1.3:5062 SIP/2.0

 

Callee:

Request-Line: BYE sip:4@192.168.1.2:5060;gr=urn:uuid:35dfa98a-2feb-482a-bde7-7568a86348b1 SIP/2.0

 

Proxy ( opensips @ 192.168.1.2 )

Status-Line: SIP/2.0 404 Not here

 

Am I missing something?

 

Should "opensipsctl ul show" show the GRUU?

 

AOR:: 4
       Contact:: sip:4@192.168.1.72:5062;transport=udp Q=
               ContactID:: 3518589640418194
               Expires:: 3586
               Callid:: OL1gvsViBJ
               Cseq:: 21
               User-agent:: LinphoneAndroid/4.0.1 (belle-sip/1.6.3)
               State:: CS_NEW
               Flags:: 0
               Cflags::  
               Socket:: udp:192.168.1.2:5060
               Methods:: 4294967295
               SIP_instance:: <urn:uuid:35dfa98a-2feb-482a-bde7-7568a86348b1>

 


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Re: GRUU contact not found

Robert Dyck

After some thought I realized that a lookup had to be invoked while in dialog. The BYE was directed at the proxy and the GRUU needed to be mapped to the device that was the intended target.

 

Added the following to script for "in dialog"

 

                       xlog("Check for GRUU, Method is $rm\n");
                       if $(ru{uri.params}{param.exist,gr}) {
                               xlog("Found GRUU\n");
                               if (!lookup("location","m")) {
                               xlog("Lookup failed\n");
                               };

 

Nov 8 11:49:30 [24807] DBG:rr:after_loose: Topmost route URI: 'sip:192.168.1.2;lr;ftag=SAjVc2sqm' is me

Nov 8 11:49:30 [24807] DBG:core:parse_headers: flags=200

Nov 8 11:49:30 [24807] DBG:core:get_hdr_field: cseq <CSeq>: <252> <BYE>

Nov 8 11:49:30 [24807] DBG:core:get_hdr_field: content_length=0

Nov 8 11:49:30 [24807] DBG:core:get_hdr_field: found end of header

Nov 8 11:49:30 [24807] DBG:rr:find_next_route: No next Route HF found

Nov 8 11:49:30 [24807] DBG:rr:after_loose: No next URI found!

Nov 8 11:49:30 [24807] DBG:core:parse_headers: flags=78

Nov 8 11:49:30 [24807] DBG:core:parse_to_param: tag=uqzwj

Nov 8 11:49:30 [24807] DBG:core:_parse_to: end of header reached, state=29

Nov 8 11:49:30 [24807] DBG:core:_parse_to: display={}, ruri={sip:7@192.168.1.2}

Nov 8 11:49:30 [24807] DBG:rr:check_route_param: params are <;lr;ftag=SAjVc2sqm>

Nov 8 11:49:30 [24807] DBG:rr:check_route_param: params are <;lr;ftag=SAjVc2sqm>

Nov 8 11:49:30 [24807] Check for GRUU, Method is BYE

Nov 8 11:49:30 [24807] Found GRUU

Nov 8 11:49:30 [24807] DBG:registrar:parse_lookup_flags: final flags: 1

Nov 8 11:49:30 [24807] DBG:registrar:extract_aor: has gruu

Nov 8 11:49:30 [24807] DBG:registrar:extract_aor: public gruu

Nov 8 11:49:30 [24807] DBG:registrar:select_contacts: ct: sip:4@192.168.1.72:5062;transport=udp

Nov 8 11:49:30 [24807] DBG:registrar:select_contacts: ruri has gruu

Nov 8 11:49:30 [24807] DBG:registrar:select_contacts: matched sip instance

Nov 8 11:49:30 [24807] DBG:registrar:push_branch: setting as ruri <sip:4@192.168.1.72:5062;transport=udp>


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Re: GRUU contact not found

Bogdan-Andrei Iancu-2
In reply to this post by Robert Dyck
Hi Robert,

According to docs, the gruu is by default off - see http://www.opensips.org/html/docs/modules/2.3.x/registrar.html#idp5567984

Regards,
Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  http://www.opensips-solutions.com
OpenSIPS Bootcamp 2018
  http://opensips.org/training/OpenSIPS_Bootcamp_2018/
On 11/07/2018 10:09 PM, Robert Dyck wrote:

My understanding is that GRUU processing in opensips is automatic, provided it is not disabled. No further configuration or scripting is required. Is that correct.

 

A GRUU capable UA rergisters and receives public and temporary GR identities. The UA establishes a dialog with another UA. The callee ends the call. The caller does not recive the BYE.

 

Caller :

Request-Line: INVITE <a class="moz-txt-link-freetext" href="sip:7@192.168.1.2">sip:7@192.168.1.2 SIP/2.0

Contact URI: <a class="moz-txt-link-freetext" href="sip:4@192.168.1.2:5060;gr=urn:uuid:35dfa98a-2feb-482a-bde7-7568a86348b1">sip:4@192.168.1.2:5060;gr=urn:uuid:35dfa98a-2feb-482a-bde7-7568a86348b1

 

Callee:

Status-Line: SIP/2.0 200 OK

 

Caller:

Request-Line: ACK <a class="moz-txt-link-freetext" href="sip:7@192.168.1.3:5062">sip:7@192.168.1.3:5062 SIP/2.0

 

Callee:

Request-Line: BYE <a class="moz-txt-link-freetext" href="sip:4@192.168.1.2:5060;gr=urn:uuid:35dfa98a-2feb-482a-bde7-7568a86348b1">sip:4@192.168.1.2:5060;gr=urn:uuid:35dfa98a-2feb-482a-bde7-7568a86348b1 SIP/2.0

 

Proxy ( opensips @ 192.168.1.2 )

Status-Line: SIP/2.0 404 Not here

 

Am I missing something?

 

Should "opensipsctl ul show" show the GRUU?

 

AOR:: 4        Contact:: <a class="moz-txt-link-freetext" href="sip:4@192.168.1.72:5062;transport=udp">sip:4@192.168.1.72:5062;transport=udp Q=                ContactID:: 3518589640418194                Expires:: 3586                Callid:: OL1gvsViBJ                Cseq:: 21                User-agent:: LinphoneAndroid/4.0.1 (belle-sip/1.6.3)                State:: CS_NEW                Flags:: 0                Cflags::                  Socket:: udp:192.168.1.2:5060                Methods:: 4294967295                SIP_instance:: <urn:uuid:35dfa98a-2feb-482a-bde7-7568a86348b1>

 



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Re: GRUU contact not found

Robert Dyck
In reply to this post by Robert Dyck

I started with the sample residential script from some time back. GRUU was enabled in the sample. When I started working with a UA that registers a GRUU I noticed it was not receiving BYE when the other end released the call. I have since experimented with checking for GRUU while in dialog. That seems to work. The documentation doesn't mention anything about modifying the script other than enabling or disabling GRUU. Do you have any tips regarding GRUU in the script? Are there corner cases I should be aware of?

 

Rob

 

On Wednesday, November 14, 2018 6:14:20 AM PST Bogdan-Andrei Iancu wrote:

Hi Robert,

According to docs, the gruu is by default off - see
http://www.opensips.org/html/docs/modules/2.3.x/registrar.html#idp5567984

Regards,

Bogdan-Andrei IancuOpenSIPS Founder and Developer http://www.opensips-solutions.comOpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/

On 11/07/2018 10:09 PM, Robert Dyck wrote:

My understanding is that GRUU processing in opensips is automatic, provided it is not disabled. No further configuration or scripting is required. Is that correct.

A GRUU capable UA rergisters and receives public and temporary GR identities. The UA establishes a dialog with another UA. The callee ends the call. The caller does not recive the BYE.

Caller :

Request-Line: INVITE <a href="sip:7@192.168.1.2">sip:7@192.168.1.2 SIP/2.0

Contact URI: <a href="sip:4@192.168.1.2:5060;gr=urn:uuid:35dfa98a-2feb-482a-bde7-7568a86348b1">sip:4@192.168.1.2:5060;gr=urn:uuid:35dfa98a-2feb-482a-bde7-7568a86348b1

Callee:

Status-Line: SIP/2.0 200 OK

Caller:

Request-Line: ACK <a href="sip:7@192.168.1.3:5062">sip:7@192.168.1.3:5062 SIP/2.0

Callee:

Request-Line: BYE <a href="sip:4@192.168.1.2:5060;gr=urn:uuid:35dfa98a-2feb-482a-bde7-7568a86348b1">sip:4@192.168.1.2:5060;gr=urn:uuid:35dfa98a-2feb-482a-bde7-7568a86348b1 SIP/2.0

Proxy ( opensips @ 192.168.1.2 )

Status-Line: SIP/2.0 404 Not here

Am I missing something?

Should "opensipsctl ul show" show the GRUU?

AOR:: 4       Contact:: <a href="sip:4@192.168.1.72:5062;transport=udp">sip:4@192.168.1.72:5062;transport=udp Q=                ContactID:: 3518589640418194                Expires:: 3586                Callid:: OL1gvsViBJ                Cseq:: 21                User-agent:: LinphoneAndroid/4.0.1 (belle-sip/1.6.3)                State:: CS_NEW                Flags:: 0                Cflags::                 Socket:: udp:192.168.1.2:5060                Methods:: 4294967295                SIP_instance:: <urn:uuid:35dfa98a-2feb-482a-bde7-7568a86348b1>



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Re: GRUU contact not found

Bogdan-Andrei Iancu-2
Hi Robert,

So you experience the issues during the sequential requests in a call, and not during the registration. Outside the registration context, the gruu does not require special support.

Could you email (off-list) a pacap showing your problem ? The pcap should be from OpenSIPS machine, showing in and out traffic, from the very beginning of the call.

Regards,
Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  http://www.opensips-solutions.com
OpenSIPS Bootcamp 2018
  http://opensips.org/training/OpenSIPS_Bootcamp_2018/
On 11/14/2018 07:09 PM, Robert Dyck wrote:

I started with the sample residential script from some time back. GRUU was enabled in the sample. When I started working with a UA that registers a GRUU I noticed it was not receiving BYE when the other end released the call. I have since experimented with checking for GRUU while in dialog. That seems to work. The documentation doesn't mention anything about modifying the script other than enabling or disabling GRUU. Do you have any tips regarding GRUU in the script? Are there corner cases I should be aware of?

 

Rob

 

On Wednesday, November 14, 2018 6:14:20 AM PST Bogdan-Andrei Iancu wrote:

Hi Robert, According to docs, the gruu is by default off - seehttp://www.opensips.org/html/docs/modules/2.3.x/registrar.html#idp5567984 Regards,

Bogdan-Andrei IancuOpenSIPS Founder and Developer http://www.opensips-solutions.comOpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/

On 11/07/2018 10:09 PM, Robert Dyck wrote:

My understanding is that GRUU processing in opensips is automatic, provided it is not disabled. No further configuration or scripting is required. Is that correct.

A GRUU capable UA rergisters and receives public and temporary GR identities. The UA establishes a dialog with another UA. The callee ends the call. The caller does not recive the BYE.

Caller :

Request-Line: INVITE <a href="sip:7@192.168.1.2" moz-do-not-send="true">sip:7@192.168.1.2 SIP/2.0

Contact URI: <a href="sip:4@192.168.1.2:5060;gr=urn:uuid:35dfa98a-2feb-482a-bde7-7568a86348b1" moz-do-not-send="true">sip:4@192.168.1.2:5060;gr=urn:uuid:35dfa98a-2feb-482a-bde7-7568a86348b1

Callee:

Status-Line: SIP/2.0 200 OK

Caller:

Request-Line: ACK <a href="sip:7@192.168.1.3:5062" moz-do-not-send="true">sip:7@192.168.1.3:5062 SIP/2.0

Callee:

Request-Line: BYE <a href="sip:4@192.168.1.2:5060;gr=urn:uuid:35dfa98a-2feb-482a-bde7-7568a86348b1" moz-do-not-send="true">sip:4@192.168.1.2:5060;gr=urn:uuid:35dfa98a-2feb-482a-bde7-7568a86348b1 SIP/2.0

Proxy ( opensips @ 192.168.1.2 )

Status-Line: SIP/2.0 404 Not here

Am I missing something?

Should "opensipsctl ul show" show the GRUU?

AOR:: 4       Contact:: <a href="sip:4@192.168.1.72:5062;transport=udp" moz-do-not-send="true">sip:4@192.168.1.72:5062;transport=udp Q=                ContactID:: 3518589640418194                Expires:: 3586                Callid:: OL1gvsViBJ                Cseq:: 21                User-agent:: LinphoneAndroid/4.0.1 (belle-sip/1.6.3)                State:: CS_NEW                Flags:: 0                Cflags::                 Socket:: udp:192.168.1.2:5060                Methods:: 4294967295                SIP_instance:: <urn:uuid:35dfa98a-2feb-482a-bde7-7568a86348b1>

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