handling multiple proxy / Record-Route

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handling multiple proxy / Record-Route

Julien Chavanton

Hi,

I have a situation whit multiple proxy where ACK is not sent as I would expect.

if we look at the following "200 OK", I am expecting ACK to be sent to 1.1.1.1 but the "Asterisk PBX 1.6.0.6." is selecting 2.2.2.2 is this normal ?

Do I have to handle Record-Route differently ?

 

 

U 1.1.1.1:5060 -> 192.168.1.108:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060.
To: <sip:[hidden email]>;tag=as664de2c2.
From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
Call-ID: [hidden email].
CSeq: 102 INVITE.
Content-Type: application/sdp.
Contact: <sip:15141234567@2.2.2.2:5060>.
Content-Length: 241.
Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
User-Agent: Packetrino.
Supported: replaces.
Record-Route: <sip:2.2.2.2:5060;lr>.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.

 

 

 

 

---------------------------------------------------------

complete SIP signaling

---------------------------------------------------------

#
U 192.168.1.108:5060 -> 1.1.1.1:5060
INVITE sip:[hidden email] SIP/2.0.
Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bK2e975bf5;rport.
Max-Forwards: 70.
From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
To: <sip:[hidden email]>.
Contact: <sip:15141234567@192.168.1.108>.
Call-ID:
[hidden email].
CSeq: 102 INVITE.
User-Agent: Asterisk PBX 1.6.0.6.
Date: Wed, 29 Apr 2009 15:38:18 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Supported: replaces, timer.
Content-Type: application/sdp.
Content-Length: 265.
.
v=0.
o=root 1992389746 1992389746 IN IP4 192.168.1.108.
s=Asterisk PBX 1.6.0.6.
c=IN IP4 192.168.1.108.
t=0 0.
m=audio 11232 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.

#
U 1.1.1.1:5060 -> 192.168.1.108:5060
SIP/2.0 100 Giving a try.
Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bK2e975bf5;rport=5060;received=74.56.45.88.
From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
To: <sip:[hidden email]>.
Call-ID:
[hidden email].
CSeq: 102 INVITE.
Server: OpenSIPS (1.4.4-notls (x86_64/linux)).
Content-Length: 0.
.

#
U 1.1.1.1:5060 -> 192.168.1.108:5060
SIP/2.0 183 Session Progress.
Via: SIP/2.0/UDP 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060.
To: <sip:[hidden email]>;tag=as664de2c2.
From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
Call-ID:
[hidden email].
CSeq: 102 INVITE.
Content-Type: application/sdp.
Contact: <sip:15141234567@2.2.2.2:5060>.
Content-Length: 241.
Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
User-Agent: Packetrino.
Supported: replaces.
Record-Route: <sip:2.2.2.2:5060;lr>.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
.
v=0.
o=root 29378 29378 IN IP4 64.2.142.160.
s=session.
c=IN IP4 1.1.1.1.
t=0 0.
m=audio 52528 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.

#
U 1.1.1.1:5060 -> 192.168.1.108:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060.
To: <sip:[hidden email]>;tag=as664de2c2.
From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
Call-ID:
[hidden email].
CSeq: 102 INVITE.
Contact: <sip:15141234567@2.2.2.2:5060>.
Content-Length: 0.
Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
User-Agent: Packetrino.
Supported: replaces.
Record-Route: <sip:2.2.2.2:5060;lr>.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
.

#
U 1.1.1.1:5060 -> 192.168.1.108:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060.
To: <sip:[hidden email]>;tag=as664de2c2.
From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
Call-ID:
[hidden email].
CSeq: 102 INVITE.
Content-Type: application/sdp.
Contact: <sip:15141234567@2.2.2.2:5060>.
Content-Length: 241.
Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
User-Agent: Packetrino.
Supported: replaces.
Record-Route: <sip:2.2.2.2:5060;lr>.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
.
v=0.
o=root 29378 29379 IN IP4 64.2.142.160.
s=session.
c=IN IP4 1.1.1.1.
t=0 0.
m=audio 52528 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.

#
U 192.168.1.108:5060 -> 2.2.2.2:5060
ACK sip:15141234567@2.2.2.2:5060 SIP/2.0.
Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bK04335252;rport.
Route: <sip:2.2.2.2:5060;lr>,<sip:1.1.1.1;lr=on;nat=yes>.
Max-Forwards: 70.
From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
To: <sip:[hidden email]>;tag=as664de2c2.
Contact: <sip:15141234567@192.168.1.108>.
Call-ID:
[hidden email].
CSeq: 102 ACK.
User-Agent: Asterisk PBX 1.6.0.6.
Content-Length: 0.
.

 

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Re: handling multiple proxy / Record-Route

Bogdan-Andrei Iancu
Hi Julien,

I think Asterisk is doing the job properly. As you see the 200 OK has:
    Contact: <sip:15141234567@2.2.2.2:5060>.
    Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
  Record-Route: <sip:2.2.2.2:5060;lr>.

So, Asterisk is generating the ACK with the Contact in RURI and the
Route set in the reverted order (correct loose routing).
    -> RURI: sip:15141234567@2.2.2.2:5060
           Destination: sip:2.2.2.2:5060;lr
     Route: sip:2.2.2.2:5060;lr + sip:1.1.1.1;lr=on;nat=yes

I think the problem here is who and why adding the bottom RR in 200 OK
(why 2 of them ?)

Regards,
Bogdan

Julien Chavanton wrote:

>
> Hi,
>
> I have a situation whit multiple proxy where ACK is not sent as I
> would expect.
>
> if we look at the following "200 OK", I am expecting ACK to be sent to
> 1.1.1.1 but the "Asterisk PBX 1.6.0.6." is selecting 2.2.2.2 is this
> normal ?
>
> Do I have to handle Record-Route differently ?
>
>  
>
>  
>
> U 1.1.1.1:5060 -> 192.168.1.108:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP
> 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060.
> To: <sip:[hidden email]>;tag=as664de2c2.
> From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> <mailto:641cab3f73fa37a871818d1a70c4061b@192.168.1.108>.
> CSeq: 102 INVITE.
> Content-Type: application/sdp.
> Contact: <sip:15141234567@2.2.2.2:5060>.
> Content-Length: 241.
> Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
> User-Agent: Packetrino.
> Supported: replaces.
> Record-Route: <sip:2.2.2.2:5060;lr>.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
>
>  
>
>  
>
>  
>
>  
>
> ---------------------------------------------------------
>
> complete SIP signaling
>
> ---------------------------------------------------------
>
> #
> U 192.168.1.108:5060 -> 1.1.1.1:5060
> INVITE sip:[hidden email] SIP/2.0.
> Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bK2e975bf5;rport.
> Max-Forwards: 70.
> From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> To: <sip:[hidden email]>.
> Contact: <sip:15141234567@192.168.1.108>.
> Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> <mailto:641cab3f73fa37a871818d1a70c4061b@192.168.1.108>.
> CSeq: 102 INVITE.
> User-Agent: Asterisk PBX 1.6.0.6.
> Date: Wed, 29 Apr 2009 15:38:18 GMT.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> Supported: replaces, timer.
> Content-Type: application/sdp.
> Content-Length: 265.
> .
> v=0.
> o=root 1992389746 1992389746 IN IP4 192.168.1.108.
> s=Asterisk PBX 1.6.0.6.
> c=IN IP4 192.168.1.108.
> t=0 0.
> m=audio 11232 RTP/AVP 0 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=silenceSupp:off - - - -.
> a=ptime:20.
> a=sendrecv.
>
> #
> U 1.1.1.1:5060 -> 192.168.1.108:5060
> SIP/2.0 100 Giving a try.
> Via: SIP/2.0/UDP
> 192.168.1.108:5060;branch=z9hG4bK2e975bf5;rport=5060;received=74.56.45.88.
> From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> To: <sip:[hidden email]>.
> Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> <mailto:641cab3f73fa37a871818d1a70c4061b@192.168.1.108>.
> CSeq: 102 INVITE.
> Server: OpenSIPS (1.4.4-notls (x86_64/linux)).
> Content-Length: 0.
> .
>
> #
> U 1.1.1.1:5060 -> 192.168.1.108:5060
> SIP/2.0 183 Session Progress.
> Via: SIP/2.0/UDP
> 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060.
> To: <sip:[hidden email]>;tag=as664de2c2.
> From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> <mailto:641cab3f73fa37a871818d1a70c4061b@192.168.1.108>.
> CSeq: 102 INVITE.
> Content-Type: application/sdp.
> Contact: <sip:15141234567@2.2.2.2:5060>.
> Content-Length: 241.
> Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
> User-Agent: Packetrino.
> Supported: replaces.
> Record-Route: <sip:2.2.2.2:5060;lr>.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> .
> v=0.
> o=root 29378 29378 IN IP4 64.2.142.160.
> s=session.
> c=IN IP4 1.1.1.1.
> t=0 0.
> m=audio 52528 RTP/AVP 0 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=silenceSupp:off - - - -.
> a=ptime:20.
> a=sendrecv.
>
> #
> U 1.1.1.1:5060 -> 192.168.1.108:5060
> SIP/2.0 180 Ringing.
> Via: SIP/2.0/UDP
> 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060.
> To: <sip:[hidden email]>;tag=as664de2c2.
> From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> <mailto:641cab3f73fa37a871818d1a70c4061b@192.168.1.108>.
> CSeq: 102 INVITE.
> Contact: <sip:15141234567@2.2.2.2:5060>.
> Content-Length: 0.
> Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
> User-Agent: Packetrino.
> Supported: replaces.
> Record-Route: <sip:2.2.2.2:5060;lr>.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> .
>
> #
> U 1.1.1.1:5060 -> 192.168.1.108:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP
> 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060.
> To: <sip:[hidden email]>;tag=as664de2c2.
> From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> <mailto:641cab3f73fa37a871818d1a70c4061b@192.168.1.108>.
> CSeq: 102 INVITE.
> Content-Type: application/sdp.
> Contact: <sip:15141234567@2.2.2.2:5060>.
> Content-Length: 241.
> Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
> User-Agent: Packetrino.
> Supported: replaces.
> Record-Route: <sip:2.2.2.2:5060;lr>.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> .
> v=0.
> o=root 29378 29379 IN IP4 64.2.142.160.
> s=session.
> c=IN IP4 1.1.1.1.
> t=0 0.
> m=audio 52528 RTP/AVP 0 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=silenceSupp:off - - - -.
> a=ptime:20.
> a=sendrecv.
>
> #
> U 192.168.1.108:5060 -> 2.2.2.2:5060
> ACK sip:15141234567@2.2.2.2:5060 SIP/2.0.
> Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bK04335252;rport.
> Route: <sip:2.2.2.2:5060;lr>,<sip:1.1.1.1;lr=on;nat=yes>.
> Max-Forwards: 70.
> From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> To: <sip:[hidden email]>;tag=as664de2c2.
> Contact: <sip:15141234567@192.168.1.108>.
> Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> <mailto:641cab3f73fa37a871818d1a70c4061b@192.168.1.108>.
> CSeq: 102 ACK.
> User-Agent: Asterisk PBX 1.6.0.6.
> Content-Length: 0.
> .
>
>  
> ------------------------------------------------------------------------
>
> _______________________________________________
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> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>  


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Re: handling multiple proxy / Record-Route

Julien Chavanton
Re: [OpenSIPS-Users] handling multiple proxy / Record-Route
 
 
UA --> PROXY 1.1.1.1 --> PROXY 2.2.2.2 --> UA
 
P1 --> P2
INVITE
Record-Route: <sip:1.1.1.1;lr=on;nat=yes>
 
P2 --> P1
100 Trying
Record-Route: <sip:1.1.1.1;lr=on;nat=yes>
Record-Route: <sip:2.2.2.2:5060;lr>
 
 
Is there something wrong ? shouldn't proxy 2.2.2.2 add his Record-Route on top of the existing Record-Route ?


From: Bogdan-Andrei Iancu [mailto:[hidden email]]
Sent: Thu 30/04/2009 8:12 AM
To: Julien Chavanton
Cc: [hidden email]
Subject: Re: [OpenSIPS-Users] handling multiple proxy / Record-Route

Hi Julien,

I think Asterisk is doing the job properly. As you see the 200 OK has:
    Contact: <sip:15141234567@2.2.2.2:5060>.
    Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
  Record-Route: <sip:2.2.2.2:5060;lr>.

So, Asterisk is generating the ACK with the Contact in RURI and the
Route set in the reverted order (correct loose routing).
    -> RURI: sip:15141234567@2.2.2.2:5060
           Destination: sip:2.2.2.2:5060;lr
     Route: sip:2.2.2.2:5060;lr + sip:1.1.1.1;lr=on;nat=yes

I think the problem here is who and why adding the bottom RR in 200 OK
(why 2 of them ?)

Regards,
Bogdan

Julien Chavanton wrote:


>
> Hi,
>
> I have a situation whit multiple proxy where ACK is not sent as I
> would expect.
>
> if we look at the following "200 OK", I am expecting ACK to be sent to
> 1.1.1.1 but the "Asterisk PBX 1.6.0.6." is selecting 2.2.2.2 is this
> normal ?
>
> Do I have to handle Record-Route differently ?
>

>

>
> U 1.1.1.1:5060 -> 192.168.1.108:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP
> 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060.
> To: <sip:[hidden email]>;tag=as664de2c2.
> From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> <[hidden email]>.
> CSeq: 102 INVITE.
> Content-Type: application/sdp.
> Contact: <sip:15141234567@2.2.2.2:5060>.
> Content-Length: 241.
> Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
> User-Agent: Packetrino.
> Supported: replaces.
> Record-Route: <sip:2.2.2.2:5060;lr>.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
>

>

>

>

>
> ---------------------------------------------------------
>
> complete SIP signaling
>
> ---------------------------------------------------------
>
> #
> U 192.168.1.108:5060 -> 1.1.1.1:5060
> INVITE sip:[hidden email] SIP/2.0.
> Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bK2e975bf5;rport.
> Max-Forwards: 70.
> From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> To: <sip:[hidden email]>.
> Contact: <sip:15141234567@192.168.1.108>.
> Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> <[hidden email]>.
> CSeq: 102 INVITE.
> User-Agent: Asterisk PBX 1.6.0.6.
> Date: Wed, 29 Apr 2009 15:38:18 GMT.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> Supported: replaces, timer.
> Content-Type: application/sdp.
> Content-Length: 265.
> .
> v=0.
> o=root 1992389746 1992389746 IN IP4 192.168.1.108.
> s=Asterisk PBX 1.6.0.6.
> c=IN IP4 192.168.1.108.
> t=0 0.
> m=audio 11232 RTP/AVP 0 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=silenceSupp:off - - - -.
> a=ptime:20.
> a=sendrecv.
>
> #
> U 1.1.1.1:5060 -> 192.168.1.108:5060
> SIP/2.0 100 Giving a try.
> Via: SIP/2.0/UDP
> 192.168.1.108:5060;branch=z9hG4bK2e975bf5;rport=5060;received=74.56.45.88.
> From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> To: <sip:[hidden email]>.
> Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> <[hidden email]>.
> CSeq: 102 INVITE.
> Server: OpenSIPS (1.4.4-notls (x86_64/linux)).
> Content-Length: 0.
> .
>
> #
> U 1.1.1.1:5060 -> 192.168.1.108:5060
> SIP/2.0 183 Session Progress.
> Via: SIP/2.0/UDP
> 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060.
> To: <sip:[hidden email]>;tag=as664de2c2.
> From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> <[hidden email]>.
> CSeq: 102 INVITE.
> Content-Type: application/sdp.
> Contact: <sip:15141234567@2.2.2.2:5060>.
> Content-Length: 241.
> Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
> User-Agent: Packetrino.
> Supported: replaces.
> Record-Route: <sip:2.2.2.2:5060;lr>.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> .
> v=0.
> o=root 29378 29378 IN IP4 64.2.142.160.
> s=session.
> c=IN IP4 1.1.1.1.
> t=0 0.
> m=audio 52528 RTP/AVP 0 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=silenceSupp:off - - - -.
> a=ptime:20.
> a=sendrecv.
>
> #
> U 1.1.1.1:5060 -> 192.168.1.108:5060
> SIP/2.0 180 Ringing.
> Via: SIP/2.0/UDP
> 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060.
> To: <sip:[hidden email]>;tag=as664de2c2.
> From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> <[hidden email]>.
> CSeq: 102 INVITE.
> Contact: <sip:15141234567@2.2.2.2:5060>.
> Content-Length: 0.
> Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
> User-Agent: Packetrino.
> Supported: replaces.
> Record-Route: <sip:2.2.2.2:5060;lr>.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> .
>
> #
> U 1.1.1.1:5060 -> 192.168.1.108:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP
> 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060.
> To: <sip:[hidden email]>;tag=as664de2c2.
> From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> <[hidden email]>.
> CSeq: 102 INVITE.
> Content-Type: application/sdp.
> Contact: <sip:15141234567@2.2.2.2:5060>.
> Content-Length: 241.
> Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
> User-Agent: Packetrino.
> Supported: replaces.
> Record-Route: <sip:2.2.2.2:5060;lr>.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> .
> v=0.
> o=root 29378 29379 IN IP4 64.2.142.160.
> s=session.
> c=IN IP4 1.1.1.1.
> t=0 0.
> m=audio 52528 RTP/AVP 0 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=silenceSupp:off - - - -.
> a=ptime:20.
> a=sendrecv.
>
> #
> U 192.168.1.108:5060 -> 2.2.2.2:5060
> ACK sip:15141234567@2.2.2.2:5060 SIP/2.0.
> Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bK04335252;rport.
> Route: <sip:2.2.2.2:5060;lr>,<sip:1.1.1.1;lr=on;nat=yes>.
> Max-Forwards: 70.
> From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> To: <sip:[hidden email]>;tag=as664de2c2.
> Contact: <sip:15141234567@192.168.1.108>.
> Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> <[hidden email]>.
> CSeq: 102 ACK.
> User-Agent: Asterisk PBX 1.6.0.6.
> Content-Length: 0.
> .
>

> ------------------------------------------------------------------------
>
> _______________________________________________
> Users mailing list
> [hidden email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>  


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Re: handling multiple proxy / Record-Route

Bogdan-Andrei Iancu
Hi Julian,

Julien Chavanton wrote:

>  
>  
> UA --> PROXY 1.1.1.1 --> PROXY 2.2.2.2 --> UA
>  
> P1 --> P2
> INVITE
> Record-Route: <sip:1.1.1.1;lr=on;nat=yes>
>  
> P2 --> P1
> 100 Trying
> Record-Route: <sip:1.1.1.1;lr=on;nat=yes>
> Record-Route: <sip:2.2.2.2:5060;lr>
>  
^^^^^^^^^^^^

This is not correct. The RR of P2 most me on top of RR of P1 - adding RR
headers works as a stack.

Regards,
Bogdan

>  
> Is there something wrong ? shouldn't proxy 2.2.2.2 add his
> Record-Route on top of the existing Record-Route ?
>
> ------------------------------------------------------------------------
> *From:* Bogdan-Andrei Iancu [mailto:[hidden email]]
> *Sent:* Thu 30/04/2009 8:12 AM
> *To:* Julien Chavanton
> *Cc:* [hidden email]
> *Subject:* Re: [OpenSIPS-Users] handling multiple proxy / Record-Route
>
> Hi Julien,
>
> I think Asterisk is doing the job properly. As you see the 200 OK has:
>     Contact: <sip:15141234567@2.2.2.2:5060>.
>     Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
>   Record-Route: <sip:2.2.2.2:5060;lr>.
>
> So, Asterisk is generating the ACK with the Contact in RURI and the
> Route set in the reverted order (correct loose routing).
>     -> RURI: sip:15141234567@2.2.2.2:5060
>            Destination: sip:2.2.2.2:5060;lr
>      Route: sip:2.2.2.2:5060;lr + sip:1.1.1.1;lr=on;nat=yes
>
> I think the problem here is who and why adding the bottom RR in 200 OK
> (why 2 of them ?)
>
> Regards,
> Bogdan
>
> Julien Chavanton wrote:
> >
> > Hi,
> >
> > I have a situation whit multiple proxy where ACK is not sent as I
> > would expect.
> >
> > if we look at the following "200 OK", I am expecting ACK to be sent to
> > 1.1.1.1 but the "Asterisk PBX 1.6.0.6." is selecting 2.2.2.2 is this
> > normal ?
> >
> > Do I have to handle Record-Route differently ?
> >
> >
> >
> >
> >
> > U 1.1.1.1:5060 -> 192.168.1.108:5060
> > SIP/2.0 200 OK.
> > Via: SIP/2.0/UDP
> >
> 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060.
> > To: <sip:[hidden email]>;tag=as664de2c2.
> > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> > <mailto:641cab3f73fa37a871818d1a70c4061b@192.168.1.108>.
> > CSeq: 102 INVITE.
> > Content-Type: application/sdp.
> > Contact: <sip:15141234567@2.2.2.2:5060>.
> > Content-Length: 241.
> > Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
> > User-Agent: Packetrino.
> > Supported: replaces.
> > Record-Route: <sip:2.2.2.2:5060;lr>.
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> >
> >
> >
> >
> >
> >
> >
> >
> >
> > ---------------------------------------------------------
> >
> > complete SIP signaling
> >
> > ---------------------------------------------------------
> >
> > #
> > U 192.168.1.108:5060 -> 1.1.1.1:5060
> > INVITE sip:[hidden email] SIP/2.0.
> > Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bK2e975bf5;rport.
> > Max-Forwards: 70.
> > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> > To: <sip:[hidden email]>.
> > Contact: <sip:15141234567@192.168.1.108>.
> > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> > <mailto:641cab3f73fa37a871818d1a70c4061b@192.168.1.108>.
> > CSeq: 102 INVITE.
> > User-Agent: Asterisk PBX 1.6.0.6.
> > Date: Wed, 29 Apr 2009 15:38:18 GMT.
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> > Supported: replaces, timer.
> > Content-Type: application/sdp.
> > Content-Length: 265.
> > .
> > v=0.
> > o=root 1992389746 1992389746 IN IP4 192.168.1.108.
> > s=Asterisk PBX 1.6.0.6.
> > c=IN IP4 192.168.1.108.
> > t=0 0.
> > m=audio 11232 RTP/AVP 0 101.
> > a=rtpmap:0 PCMU/8000.
> > a=rtpmap:101 telephone-event/8000.
> > a=fmtp:101 0-16.
> > a=silenceSupp:off - - - -.
> > a=ptime:20.
> > a=sendrecv.
> >
> > #
> > U 1.1.1.1:5060 -> 192.168.1.108:5060
> > SIP/2.0 100 Giving a try.
> > Via: SIP/2.0/UDP
> >
> 192.168.1.108:5060;branch=z9hG4bK2e975bf5;rport=5060;received=74.56.45.88.
> > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> > To: <sip:[hidden email]>.
> > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> > <mailto:641cab3f73fa37a871818d1a70c4061b@192.168.1.108>.
> > CSeq: 102 INVITE.
> > Server: OpenSIPS (1.4.4-notls (x86_64/linux)).
> > Content-Length: 0.
> > .
> >
> > #
> > U 1.1.1.1:5060 -> 192.168.1.108:5060
> > SIP/2.0 183 Session Progress.
> > Via: SIP/2.0/UDP
> >
> 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060.
> > To: <sip:[hidden email]>;tag=as664de2c2.
> > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> > <mailto:641cab3f73fa37a871818d1a70c4061b@192.168.1.108>.
> > CSeq: 102 INVITE.
> > Content-Type: application/sdp.
> > Contact: <sip:15141234567@2.2.2.2:5060>.
> > Content-Length: 241.
> > Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
> > User-Agent: Packetrino.
> > Supported: replaces.
> > Record-Route: <sip:2.2.2.2:5060;lr>.
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> > .
> > v=0.
> > o=root 29378 29378 IN IP4 64.2.142.160.
> > s=session.
> > c=IN IP4 1.1.1.1.
> > t=0 0.
> > m=audio 52528 RTP/AVP 0 101.
> > a=rtpmap:0 PCMU/8000.
> > a=rtpmap:101 telephone-event/8000.
> > a=fmtp:101 0-16.
> > a=silenceSupp:off - - - -.
> > a=ptime:20.
> > a=sendrecv.
> >
> > #
> > U 1.1.1.1:5060 -> 192.168.1.108:5060
> > SIP/2.0 180 Ringing.
> > Via: SIP/2.0/UDP
> >
> 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060.
> > To: <sip:[hidden email]>;tag=as664de2c2.
> > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> > <mailto:641cab3f73fa37a871818d1a70c4061b@192.168.1.108>.
> > CSeq: 102 INVITE.
> > Contact: <sip:15141234567@2.2.2.2:5060>.
> > Content-Length: 0.
> > Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
> > User-Agent: Packetrino.
> > Supported: replaces.
> > Record-Route: <sip:2.2.2.2:5060;lr>.
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> > .
> >
> > #
> > U 1.1.1.1:5060 -> 192.168.1.108:5060
> > SIP/2.0 200 OK.
> > Via: SIP/2.0/UDP
> >
> 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060.
> > To: <sip:[hidden email]>;tag=as664de2c2.
> > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> > <mailto:641cab3f73fa37a871818d1a70c4061b@192.168.1.108>.
> > CSeq: 102 INVITE.
> > Content-Type: application/sdp.
> > Contact: <sip:15141234567@2.2.2.2:5060>.
> > Content-Length: 241.
> > Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
> > User-Agent: Packetrino.
> > Supported: replaces.
> > Record-Route: <sip:2.2.2.2:5060;lr>.
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> > .
> > v=0.
> > o=root 29378 29379 IN IP4 64.2.142.160.
> > s=session.
> > c=IN IP4 1.1.1.1.
> > t=0 0.
> > m=audio 52528 RTP/AVP 0 101.
> > a=rtpmap:0 PCMU/8000.
> > a=rtpmap:101 telephone-event/8000.
> > a=fmtp:101 0-16.
> > a=silenceSupp:off - - - -.
> > a=ptime:20.
> > a=sendrecv.
> >
> > #
> > U 192.168.1.108:5060 -> 2.2.2.2:5060
> > ACK sip:15141234567@2.2.2.2:5060 SIP/2.0.
> > Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bK04335252;rport.
> > Route: <sip:2.2.2.2:5060;lr>,<sip:1.1.1.1;lr=on;nat=yes>.
> > Max-Forwards: 70.
> > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> > To: <sip:[hidden email]>;tag=as664de2c2.
> > Contact: <sip:15141234567@192.168.1.108>.
> > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> > <mailto:641cab3f73fa37a871818d1a70c4061b@192.168.1.108>.
> > CSeq: 102 ACK.
> > User-Agent: Asterisk PBX 1.6.0.6.
> > Content-Length: 0.
> > .
> >
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > Users mailing list
> > [hidden email]
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >  
>


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Re: handling multiple proxy / Record-Route

Julien Chavanton
Re: [OpenSIPS-Users] handling multiple proxy / Record-Route
thank you, this is a problem as I do not control this proxy (2.2.2.2), is there a suggested way of handling this problem ?
 
Maybe there is something esle wrong on my side cusaing this problem so I am including the SIP communication between the proxy this time
 
 
 
#
U 1.1.1.1:5060 -> 2.2.2.2:5060
INVITE sip:15148622633@2.2.2.2 SIP/2.0.
Record-Route: <sip:1.1.1.1;lr>.
Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bK09e6.36a0f975.0.
Via: SIP/2.0/UDP 10.0.1.74:58366;received=10.0.1.74;branch=z9hG4bK-d87543-0f348609f47bda44-1--d87543-;rport=58366.
Max-Forwards: 69.
Contact: <sip:777@10.0.1.74:58366>.
To: "15141234567"<sip:[hidden email]>.
From: "777"<sip:[hidden email]>;tag=a030735d.
Call-ID: 8116f933cc4ea03fMjYzN2Q1MGQ5Y2M1ZDc5Yzk4OTRjN2Y5YzEwYWMwMzc..
CSeq: 1 INVITE.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO.
Content-Type: application/sdp.
User-Agent: eyeBeam release 1003s stamp 31159.
Content-Length: 478.
P-hint: Route[6]: mediaproxy .
.
v=0.
o=- 8 2 IN IP4 10.0.1.74.
s=CounterPath eyeBeam 1.5.
c=IN IP4 1.1.1.1.
t=0 0.
m=audio 52550 RTP/AVP 0 8 18 101.
a=alt:1 4 : LM6OZaAl 4x8r9qea 192.168.1.101 50006.
a=alt:2 3 : 84SVypDj oi4PbxZ7 192.168.114.1 50006.
a=alt:3 2 : L4wf6+MH s4gK5GAV 192.168.146.1 50006.
a=alt:4 1 : cg2pbkCG WDFvj29+ 10.0.1.74 50006.
a=fmtp:18 annexb=no.
a=fmtp:101 0-15.
a=rtpmap:101 telephone-event/8000.
a=sendrecv.
a=x-rtp-session-id:D56BCBC26473491FA111854E4C9F3575.
a=direction:active.
#
U 2.2.2.2:5060 -> 1.1.1.1:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK09e6.36a0f975.0;received=1.1.1.1;rport=5060.
Via: SIP/2.0/UDP 10.0.1.74:58366;received=10.0.1.74;branch=z9hG4bK-d87543-0f348609f47bda44-1--d87543-;rport=58366.
To: "15141234567" <sip:[hidden email]>.
From: "777" <sip:[hidden email]>;tag=a030735d.
Call-ID: 8116f933cc4ea03fMjYzN2Q1MGQ5Y2M1ZDc5Yzk4OTRjN2Y5YzEwYWMwMzc..
CSeq: 1 INVITE.
Contact: <sip:15148622633@64.2.142.75>.
Content-Length: 0.
Record-Route: <sip:1.1.1.1;lr>.
User-Agent: Packetrino.
Supported: replaces.
Record-Route: <sip:2.2.2.2:5060;lr>.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
.
 


From: Bogdan-Andrei Iancu [mailto:[hidden email]]
Sent: Thu 30/04/2009 3:44 PM
To: Julien Chavanton
Cc: [hidden email]
Subject: Re: [OpenSIPS-Users] handling multiple proxy / Record-Route

Hi Julian,

Julien Chavanton wrote:




> UA --> PROXY 1.1.1.1 --> PROXY 2.2.2.2 --> UA

> P1 --> P2
> INVITE
> Record-Route: <sip:1.1.1.1;lr=on;nat=yes>

> P2 --> P1
> 100 Trying
> Record-Route: <sip:1.1.1.1;lr=on;nat=yes>
> Record-Route: <sip:2.2.2.2:5060;lr>
^^^^^^^^^^^^

This is not correct. The RR of P2 most me on top of RR of P1 - adding RR
headers works as a stack.

Regards,
Bogdan


> Is there something wrong ? shouldn't proxy 2.2.2.2 add his
> Record-Route on top of the existing Record-Route ?
>
> ------------------------------------------------------------------------
> *From:* Bogdan-Andrei Iancu [[hidden email]]
> *Sent:* Thu 30/04/2009 8:12 AM
> *To:* Julien Chavanton
> *Cc:* [hidden email]
> *Subject:* Re: [OpenSIPS-Users] handling multiple proxy / Record-Route
>
> Hi Julien,
>
> I think Asterisk is doing the job properly. As you see the 200 OK has:
>     Contact: <sip:15141234567@2.2.2.2:5060>.
>     Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
>   Record-Route: <sip:2.2.2.2:5060;lr>.
>
> So, Asterisk is generating the ACK with the Contact in RURI and the
> Route set in the reverted order (correct loose routing).
>     -> RURI: sip:15141234567@2.2.2.2:5060
>            Destination: sip:2.2.2.2:5060;lr
>      Route: sip:2.2.2.2:5060;lr + sip:1.1.1.1;lr=on;nat=yes
>
> I think the problem here is who and why adding the bottom RR in 200 OK
> (why 2 of them ?)
>
> Regards,
> Bogdan
>
> Julien Chavanton wrote:
> >
> > Hi,
> >
> > I have a situation whit multiple proxy where ACK is not sent as I
> > would expect.
> >
> > if we look at the following "200 OK", I am expecting ACK to be sent to
> > 1.1.1.1 but the "Asterisk PBX 1.6.0.6." is selecting 2.2.2.2 is this
> > normal ?
> >
> > Do I have to handle Record-Route differently ?
> >
> >
> >
> >
> >
> > U 1.1.1.1:5060 -> 192.168.1.108:5060
> > SIP/2.0 200 OK.
> > Via: SIP/2.0/UDP
> >
> 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060.
> > To: <sip:[hidden email]>;tag=as664de2c2.
> > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> > <[hidden email]>.
> > CSeq: 102 INVITE.
> > Content-Type: application/sdp.
> > Contact: <sip:15141234567@2.2.2.2:5060>.
> > Content-Length: 241.
> > Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
> > User-Agent: Packetrino.
> > Supported: replaces.
> > Record-Route: <sip:2.2.2.2:5060;lr>.
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> >
> >
> >
> >
> >
> >
> >
> >
> >
> > ---------------------------------------------------------
> >
> > complete SIP signaling
> >
> > ---------------------------------------------------------
> >
> > #
> > U 192.168.1.108:5060 -> 1.1.1.1:5060
> > INVITE sip:[hidden email] SIP/2.0.
> > Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bK2e975bf5;rport.
> > Max-Forwards: 70.
> > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> > To: <sip:[hidden email]>.
> > Contact: <sip:15141234567@192.168.1.108>.
> > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> > <[hidden email]>.
> > CSeq: 102 INVITE.
> > User-Agent: Asterisk PBX 1.6.0.6.
> > Date: Wed, 29 Apr 2009 15:38:18 GMT.
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> > Supported: replaces, timer.
> > Content-Type: application/sdp.
> > Content-Length: 265.
> > .
> > v=0.
> > o=root 1992389746 1992389746 IN IP4 192.168.1.108.
> > s=Asterisk PBX 1.6.0.6.
> > c=IN IP4 192.168.1.108.
> > t=0 0.
> > m=audio 11232 RTP/AVP 0 101.
> > a=rtpmap:0 PCMU/8000.
> > a=rtpmap:101 telephone-event/8000.
> > a=fmtp:101 0-16.
> > a=silenceSupp:off - - - -.
> > a=ptime:20.
> > a=sendrecv.
> >
> > #
> > U 1.1.1.1:5060 -> 192.168.1.108:5060
> > SIP/2.0 100 Giving a try.
> > Via: SIP/2.0/UDP
> >
> 192.168.1.108:5060;branch=z9hG4bK2e975bf5;rport=5060;received=74.56.45.88.
> > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> > To: <sip:[hidden email]>.
> > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> > <[hidden email]>.
> > CSeq: 102 INVITE.
> > Server: OpenSIPS (1.4.4-notls (x86_64/linux)).
> > Content-Length: 0.
> > .
> >
> > #
> > U 1.1.1.1:5060 -> 192.168.1.108:5060
> > SIP/2.0 183 Session Progress.
> > Via: SIP/2.0/UDP
> >
> 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060.
> > To: <sip:[hidden email]>;tag=as664de2c2.
> > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> > <[hidden email]>.
> > CSeq: 102 INVITE.
> > Content-Type: application/sdp.
> > Contact: <sip:15141234567@2.2.2.2:5060>.
> > Content-Length: 241.
> > Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
> > User-Agent: Packetrino.
> > Supported: replaces.
> > Record-Route: <sip:2.2.2.2:5060;lr>.
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> > .
> > v=0.
> > o=root 29378 29378 IN IP4 64.2.142.160.
> > s=session.
> > c=IN IP4 1.1.1.1.
> > t=0 0.
> > m=audio 52528 RTP/AVP 0 101.
> > a=rtpmap:0 PCMU/8000.
> > a=rtpmap:101 telephone-event/8000.
> > a=fmtp:101 0-16.
> > a=silenceSupp:off - - - -.
> > a=ptime:20.
> > a=sendrecv.
> >
> > #
> > U 1.1.1.1:5060 -> 192.168.1.108:5060
> > SIP/2.0 180 Ringing.
> > Via: SIP/2.0/UDP
> >
> 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060.
> > To: <sip:[hidden email]>;tag=as664de2c2.
> > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> > <[hidden email]>.
> > CSeq: 102 INVITE.
> > Contact: <sip:15141234567@2.2.2.2:5060>.
> > Content-Length: 0.
> > Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
> > User-Agent: Packetrino.
> > Supported: replaces.
> > Record-Route: <sip:2.2.2.2:5060;lr>.
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> > .
> >
> > #
> > U 1.1.1.1:5060 -> 192.168.1.108:5060
> > SIP/2.0 200 OK.
> > Via: SIP/2.0/UDP
> >
> 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060.
> > To: <sip:[hidden email]>;tag=as664de2c2.
> > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> > <[hidden email]>.
> > CSeq: 102 INVITE.
> > Content-Type: application/sdp.
> > Contact: <sip:15141234567@2.2.2.2:5060>.
> > Content-Length: 241.
> > Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
> > User-Agent: Packetrino.
> > Supported: replaces.
> > Record-Route: <sip:2.2.2.2:5060;lr>.
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> > .
> > v=0.
> > o=root 29378 29379 IN IP4 64.2.142.160.
> > s=session.
> > c=IN IP4 1.1.1.1.
> > t=0 0.
> > m=audio 52528 RTP/AVP 0 101.
> > a=rtpmap:0 PCMU/8000.
> > a=rtpmap:101 telephone-event/8000.
> > a=fmtp:101 0-16.
> > a=silenceSupp:off - - - -.
> > a=ptime:20.
> > a=sendrecv.
> >
> > #
> > U 192.168.1.108:5060 -> 2.2.2.2:5060
> > ACK sip:15141234567@2.2.2.2:5060 SIP/2.0.
> > Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bK04335252;rport.
> > Route: <sip:2.2.2.2:5060;lr>,<sip:1.1.1.1;lr=on;nat=yes>.
> > Max-Forwards: 70.
> > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> > To: <sip:[hidden email]>;tag=as664de2c2.
> > Contact: <sip:15141234567@192.168.1.108>.
> > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> > <[hidden email]>.
> > CSeq: 102 ACK.
> > User-Agent: Asterisk PBX 1.6.0.6.
> > Content-Length: 0.
> > .
> >
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > Users mailing list
> > [hidden email]
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> > 
>


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Re: handling multiple proxy / Record-Route

Julien Chavanton
Re: [OpenSIPS-Users] handling multiple proxy / Record-Route
I think I will try the option to use the "textops" module to enforce the correct order of Record-Route to validate this is my problem etc.
 
 


From: [hidden email] on behalf of Julien Chavanton
Sent: Thu 30/04/2009 3:44 PM
To: Bogdan-Andrei Iancu
Cc: [hidden email]
Subject: Re: [OpenSIPS-Users] handling multiple proxy / Record-Route

thank you, this is a problem as I do not control this proxy (2.2.2.2), is there a suggested way of handling this problem ?
 
Maybe there is something esle wrong on my side cusaing this problem so I am including the SIP communication between the proxy this time
 
 
 
#
U 1.1.1.1:5060 -> 2.2.2.2:5060
INVITE sip:15148622633@2.2.2.2 SIP/2.0.
Record-Route: <sip:1.1.1.1;lr>.
Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bK09e6.36a0f975.0.
Via: SIP/2.0/UDP 10.0.1.74:58366;received=10.0.1.74;branch=z9hG4bK-d87543-0f348609f47bda44-1--d87543-;rport=58366.
Max-Forwards: 69.
Contact: <sip:777@10.0.1.74:58366>.
To: "15141234567"<sip:[hidden email]>.
From: "777"<sip:[hidden email]>;tag=a030735d.
Call-ID: 8116f933cc4ea03fMjYzN2Q1MGQ5Y2M1ZDc5Yzk4OTRjN2Y5YzEwYWMwMzc..
CSeq: 1 INVITE.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO.
Content-Type: application/sdp.
User-Agent: eyeBeam release 1003s stamp 31159.
Content-Length: 478.
P-hint: Route[6]: mediaproxy .
.
v=0.
o=- 8 2 IN IP4 10.0.1.74.
s=CounterPath eyeBeam 1.5.
c=IN IP4 1.1.1.1.
t=0 0.
m=audio 52550 RTP/AVP 0 8 18 101.
a=alt:1 4 : LM6OZaAl 4x8r9qea 192.168.1.101 50006.
a=alt:2 3 : 84SVypDj oi4PbxZ7 192.168.114.1 50006.
a=alt:3 2 : L4wf6+MH s4gK5GAV 192.168.146.1 50006.
a=alt:4 1 : cg2pbkCG WDFvj29+ 10.0.1.74 50006.
a=fmtp:18 annexb=no.
a=fmtp:101 0-15.
a=rtpmap:101 telephone-event/8000.
a=sendrecv.
a=x-rtp-session-id:D56BCBC26473491FA111854E4C9F3575.
a=direction:active.
#
U 2.2.2.2:5060 -> 1.1.1.1:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK09e6.36a0f975.0;received=1.1.1.1;rport=5060.
Via: SIP/2.0/UDP 10.0.1.74:58366;received=10.0.1.74;branch=z9hG4bK-d87543-0f348609f47bda44-1--d87543-;rport=58366.
To: "15141234567" <sip:[hidden email]>.
From: "777" <sip:[hidden email]>;tag=a030735d.
Call-ID: 8116f933cc4ea03fMjYzN2Q1MGQ5Y2M1ZDc5Yzk4OTRjN2Y5YzEwYWMwMzc..
CSeq: 1 INVITE.
Contact: <sip:15141234567@2.2.2.2>.
Content-Length: 0.
Record-Route: <sip:1.1.1.1;lr>.
User-Agent: Packetrino.
Supported: replaces.
Record-Route: <sip:2.2.2.2:5060;lr>.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
.
 


From: Bogdan-Andrei Iancu [mailto:[hidden email]]
Sent: Thu 30/04/2009 3:44 PM
To: Julien Chavanton
Cc: [hidden email]
Subject: Re: [OpenSIPS-Users] handling multiple proxy / Record-Route

Hi Julian,

Julien Chavanton wrote:




> UA --> PROXY 1.1.1.1 --> PROXY 2.2.2.2 --> UA

> P1 --> P2
> INVITE
> Record-Route: <sip:1.1.1.1;lr=on;nat=yes>

> P2 --> P1
> 100 Trying
> Record-Route: <sip:1.1.1.1;lr=on;nat=yes>
> Record-Route: <sip:2.2.2.2:5060;lr>
^^^^^^^^^^^^

This is not correct. The RR of P2 most me on top of RR of P1 - adding RR
headers works as a stack.

Regards,
Bogdan


> Is there something wrong ? shouldn't proxy 2.2.2.2 add his
> Record-Route on top of the existing Record-Route ?
>
> ------------------------------------------------------------------------
> *From:* Bogdan-Andrei Iancu [[hidden email]]
> *Sent:* Thu 30/04/2009 8:12 AM
> *To:* Julien Chavanton
> *Cc:* [hidden email]
> *Subject:* Re: [OpenSIPS-Users] handling multiple proxy / Record-Route
>
> Hi Julien,
>
> I think Asterisk is doing the job properly. As you see the 200 OK has:
>     Contact: <sip:15141234567@2.2.2.2:5060>.
>     Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
>   Record-Route: <sip:2.2.2.2:5060;lr>.
>
> So, Asterisk is generating the ACK with the Contact in RURI and the
> Route set in the reverted order (correct loose routing).
>     -> RURI: sip:15141234567@2.2.2.2:5060
>            Destination: sip:2.2.2.2:5060;lr
>      Route: sip:2.2.2.2:5060;lr + sip:1.1.1.1;lr=on;nat=yes
>
> I think the problem here is who and why adding the bottom RR in 200 OK
> (why 2 of them ?)
>
> Regards,
> Bogdan
>
> Julien Chavanton wrote:
> >
> > Hi,
> >
> > I have a situation whit multiple proxy where ACK is not sent as I
> > would expect.
> >
> > if we look at the following "200 OK", I am expecting ACK to be sent to
> > 1.1.1.1 but the "Asterisk PBX 1.6.0.6." is selecting 2.2.2.2 is this
> > normal ?
> >
> > Do I have to handle Record-Route differently ?
> >
> >
> >
> >
> >
> > U 1.1.1.1:5060 -> 192.168.1.108:5060
> > SIP/2.0 200 OK.
> > Via: SIP/2.0/UDP
> >
> 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060.
> > To: <sip:[hidden email]>;tag=as664de2c2.
> > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> > <[hidden email]>.
> > CSeq: 102 INVITE.
> > Content-Type: application/sdp.
> > Contact: <sip:15141234567@2.2.2.2:5060>.
> > Content-Length: 241.
> > Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
> > User-Agent: Packetrino.
> > Supported: replaces.
> > Record-Route: <sip:2.2.2.2:5060;lr>.
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> >
> >
> >
> >
> >
> >
> >
> >
> >
> > ---------------------------------------------------------
> >
> > complete SIP signaling
> >
> > ---------------------------------------------------------
> >
> > #
> > U 192.168.1.108:5060 -> 1.1.1.1:5060
> > INVITE sip:[hidden email] SIP/2.0.
> > Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bK2e975bf5;rport.
> > Max-Forwards: 70.
> > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> > To: <sip:[hidden email]>.
> > Contact: <sip:15141234567@192.168.1.108>.
> > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> > <[hidden email]>.
> > CSeq: 102 INVITE.
> > User-Agent: Asterisk PBX 1.6.0.6.
> > Date: Wed, 29 Apr 2009 15:38:18 GMT.
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> > Supported: replaces, timer.
> > Content-Type: application/sdp.
> > Content-Length: 265.
> > .
> > v=0.
> > o=root 1992389746 1992389746 IN IP4 192.168.1.108.
> > s=Asterisk PBX 1.6.0.6.
> > c=IN IP4 192.168.1.108.
> > t=0 0.
> > m=audio 11232 RTP/AVP 0 101.
> > a=rtpmap:0 PCMU/8000.
> > a=rtpmap:101 telephone-event/8000.
> > a=fmtp:101 0-16.
> > a=silenceSupp:off - - - -.
> > a=ptime:20.
> > a=sendrecv.
> >
> > #
> > U 1.1.1.1:5060 -> 192.168.1.108:5060
> > SIP/2.0 100 Giving a try.
> > Via: SIP/2.0/UDP
> >
> 192.168.1.108:5060;branch=z9hG4bK2e975bf5;rport=5060;received=74.56.45.88.
> > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> > To: <sip:[hidden email]>.
> > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> > <[hidden email]>.
> > CSeq: 102 INVITE.
> > Server: OpenSIPS (1.4.4-notls (x86_64/linux)).
> > Content-Length: 0.
> > .
> >
> > #
> > U 1.1.1.1:5060 -> 192.168.1.108:5060
> > SIP/2.0 183 Session Progress.
> > Via: SIP/2.0/UDP
> >
> 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060.
> > To: <sip:[hidden email]>;tag=as664de2c2.
> > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> > <[hidden email]>.
> > CSeq: 102 INVITE.
> > Content-Type: application/sdp.
> > Contact: <sip:15141234567@2.2.2.2:5060>.
> > Content-Length: 241.
> > Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
> > User-Agent: Packetrino.
> > Supported: replaces.
> > Record-Route: <sip:2.2.2.2:5060;lr>.
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> > .
> > v=0.
> > o=root 29378 29378 IN IP4 64.2.142.160.
> > s=session.
> > c=IN IP4 1.1.1.1.
> > t=0 0.
> > m=audio 52528 RTP/AVP 0 101.
> > a=rtpmap:0 PCMU/8000.
> > a=rtpmap:101 telephone-event/8000.
> > a=fmtp:101 0-16.
> > a=silenceSupp:off - - - -.
> > a=ptime:20.
> > a=sendrecv.
> >
> > #
> > U 1.1.1.1:5060 -> 192.168.1.108:5060
> > SIP/2.0 180 Ringing.
> > Via: SIP/2.0/UDP
> >
> 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060.
> > To: <sip:[hidden email]>;tag=as664de2c2.
> > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> > <[hidden email]>.
> > CSeq: 102 INVITE.
> > Contact: <sip:15141234567@2.2.2.2:5060>.
> > Content-Length: 0.
> > Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
> > User-Agent: Packetrino.
> > Supported: replaces.
> > Record-Route: <sip:2.2.2.2:5060;lr>.
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> > .
> >
> > #
> > U 1.1.1.1:5060 -> 192.168.1.108:5060
> > SIP/2.0 200 OK.
> > Via: SIP/2.0/UDP
> >
> 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060.
> > To: <sip:[hidden email]>;tag=as664de2c2.
> > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> > <[hidden email]>.
> > CSeq: 102 INVITE.
> > Content-Type: application/sdp.
> > Contact: <sip:15141234567@2.2.2.2:5060>.
> > Content-Length: 241.
> > Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
> > User-Agent: Packetrino.
> > Supported: replaces.
> > Record-Route: <sip:2.2.2.2:5060;lr>.
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> > .
> > v=0.
> > o=root 29378 29379 IN IP4 64.2.142.160.
> > s=session.
> > c=IN IP4 1.1.1.1.
> > t=0 0.
> > m=audio 52528 RTP/AVP 0 101.
> > a=rtpmap:0 PCMU/8000.
> > a=rtpmap:101 telephone-event/8000.
> > a=fmtp:101 0-16.
> > a=silenceSupp:off - - - -.
> > a=ptime:20.
> > a=sendrecv.
> >
> > #
> > U 192.168.1.108:5060 -> 2.2.2.2:5060
> > ACK sip:15141234567@2.2.2.2:5060 SIP/2.0.
> > Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bK04335252;rport.
> > Route: <sip:2.2.2.2:5060;lr>,<sip:1.1.1.1;lr=on;nat=yes>.
> > Max-Forwards: 70.
> > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> > To: <sip:[hidden email]>;tag=as664de2c2.
> > Contact: <sip:15141234567@192.168.1.108>.
> > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> > <[hidden email]>.
> > CSeq: 102 ACK.
> > User-Agent: Asterisk PBX 1.6.0.6.
> > Content-Length: 0.
> > .
> >
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > Users mailing list
> > [hidden email]
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> > 
>


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Re: handling multiple proxy / Record-Route

Bogdan-Andrei Iancu
In reply to this post by Julien Chavanton
Hi Julien,

as the bogus proxy is the last on the path (just before the client), it
is not much you can do about.

Even if you try to fix the order in the 200 OK reply, it will not work
(only partial) as the callee will still have the bogus order, so it will
not be able to route to the caller.

Regards,
Bogdan

Julien Chavanton wrote:

> thank you, this is a problem as I do not control this proxy (2.2.2.2),
> is there a suggested way of handling this problem ?
>  
> Maybe there is something esle wrong on my side cusaing this problem so
> I am including the SIP communication between the proxy this time
>  
>  
>  
> #
> U 1.1.1.1:5060 -> 2.2.2.2:5060
> INVITE sip:15148622633@2.2.2.2 SIP/2.0.
> Record-Route: <sip:1.1.1.1;lr>.
> Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bK09e6.36a0f975.0.
> Via: SIP/2.0/UDP
> 10.0.1.74:58366;received=10.0.1.74;branch=z9hG4bK-d87543-0f348609f47bda44-1--d87543-;rport=58366.
> Max-Forwards: 69.
> Contact: <sip:777@10.0.1.74:58366>.
> To: "15141234567"<sip:[hidden email]>.
> From: "777"<sip:[hidden email]>;tag=a030735d.
> Call-ID: 8116f933cc4ea03fMjYzN2Q1MGQ5Y2M1ZDc5Yzk4OTRjN2Y5YzEwYWMwMzc..
> CSeq: 1 INVITE.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
> SUBSCRIBE, INFO.
> Content-Type: application/sdp.
> User-Agent: eyeBeam release 1003s stamp 31159.
> Content-Length: 478.
> P-hint: Route[6]: mediaproxy .
> .
> v=0.
> o=- 8 2 IN IP4 10.0.1.74.
> s=CounterPath eyeBeam 1.5.
> c=IN IP4 1.1.1.1.
> t=0 0.
> m=audio 52550 RTP/AVP 0 8 18 101.
> a=alt:1 4 : LM6OZaAl 4x8r9qea 192.168.1.101 50006.
> a=alt:2 3 : 84SVypDj oi4PbxZ7 192.168.114.1 50006.
> a=alt:3 2 : L4wf6+MH s4gK5GAV 192.168.146.1 50006.
> a=alt:4 1 : cg2pbkCG WDFvj29+ 10.0.1.74 50006.
> a=fmtp:18 annexb=no.
> a=fmtp:101 0-15.
> a=rtpmap:101 telephone-event/8000.
> a=sendrecv.
> a=x-rtp-session-id:D56BCBC26473491FA111854E4C9F3575.
> a=direction:active.
> #
> U 2.2.2.2:5060 -> 1.1.1.1:5060
> SIP/2.0 100 Trying.
> Via: SIP/2.0/UDP
> 1.1.1.1:5060;branch=z9hG4bK09e6.36a0f975.0;received=1.1.1.1;rport=5060.
> Via: SIP/2.0/UDP
> 10.0.1.74:58366;received=10.0.1.74;branch=z9hG4bK-d87543-0f348609f47bda44-1--d87543-;rport=58366.
> To: "15141234567" <sip:[hidden email]>.
> From: "777" <sip:[hidden email]>;tag=a030735d.
> Call-ID: 8116f933cc4ea03fMjYzN2Q1MGQ5Y2M1ZDc5Yzk4OTRjN2Y5YzEwYWMwMzc..
> CSeq: 1 INVITE.
> Contact: <sip:15148622633@64.2.142.75>.
> Content-Length: 0.
> Record-Route: <sip:1.1.1.1;lr>.
> User-Agent: Packetrino.
> Supported: replaces.
> Record-Route: <sip:2.2.2.2:5060;lr>.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> .
>  
>
> ------------------------------------------------------------------------
> *From:* Bogdan-Andrei Iancu [mailto:[hidden email]]
> *Sent:* Thu 30/04/2009 3:44 PM
> *To:* Julien Chavanton
> *Cc:* [hidden email]
> *Subject:* Re: [OpenSIPS-Users] handling multiple proxy / Record-Route
>
> Hi Julian,
>
> Julien Chavanton wrote:
> >
> >
> > UA --> PROXY 1.1.1.1 --> PROXY 2.2.2.2 --> UA
> >
> > P1 --> P2
> > INVITE
> > Record-Route: <sip:1.1.1.1;lr=on;nat=yes>
> >
> > P2 --> P1
> > 100 Trying
> > Record-Route: <sip:1.1.1.1;lr=on;nat=yes>
> > Record-Route: <sip:2.2.2.2:5060;lr>
> >
> ^^^^^^^^^^^^
>
> This is not correct. The RR of P2 most me on top of RR of P1 - adding RR
> headers works as a stack.
>
> Regards,
> Bogdan
> >
> > Is there something wrong ? shouldn't proxy 2.2.2.2 add his
> > Record-Route on top of the existing Record-Route ?
> >
> > ------------------------------------------------------------------------
> > *From:* Bogdan-Andrei Iancu [mailto:[hidden email]]
> > *Sent:* Thu 30/04/2009 8:12 AM
> > *To:* Julien Chavanton
> > *Cc:* [hidden email]
> > *Subject:* Re: [OpenSIPS-Users] handling multiple proxy / Record-Route
> >
> > Hi Julien,
> >
> > I think Asterisk is doing the job properly. As you see the 200 OK has:
> >     Contact: <sip:15141234567@2.2.2.2:5060>.
> >     Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
> >   Record-Route: <sip:2.2.2.2:5060;lr>.
> >
> > So, Asterisk is generating the ACK with the Contact in RURI and the
> > Route set in the reverted order (correct loose routing).
> >     -> RURI: sip:15141234567@2.2.2.2:5060
> >            Destination: sip:2.2.2.2:5060;lr
> >      Route: sip:2.2.2.2:5060;lr + sip:1.1.1.1;lr=on;nat=yes
> >
> > I think the problem here is who and why adding the bottom RR in 200 OK
> > (why 2 of them ?)
> >
> > Regards,
> > Bogdan
> >
> > Julien Chavanton wrote:
> > >
> > > Hi,
> > >
> > > I have a situation whit multiple proxy where ACK is not sent as I
> > > would expect.
> > >
> > > if we look at the following "200 OK", I am expecting ACK to be sent to
> > > 1.1.1.1 but the "Asterisk PBX 1.6.0.6." is selecting 2.2.2.2 is this
> > > normal ?
> > >
> > > Do I have to handle Record-Route differently ?
> > >
> > >
> > >
> > >
> > >
> > > U 1.1.1.1:5060 -> 192.168.1.108:5060
> > > SIP/2.0 200 OK.
> > > Via: SIP/2.0/UDP
> > >
> >
> 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060.
> > > To: <sip:[hidden email]>;tag=as664de2c2.
> > > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> > > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> > > <mailto:641cab3f73fa37a871818d1a70c4061b@192.168.1.108>.
> > > CSeq: 102 INVITE.
> > > Content-Type: application/sdp.
> > > Contact: <sip:15141234567@2.2.2.2:5060>.
> > > Content-Length: 241.
> > > Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
> > > User-Agent: Packetrino.
> > > Supported: replaces.
> > > Record-Route: <sip:2.2.2.2:5060;lr>.
> > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > > ---------------------------------------------------------
> > >
> > > complete SIP signaling
> > >
> > > ---------------------------------------------------------
> > >
> > > #
> > > U 192.168.1.108:5060 -> 1.1.1.1:5060
> > > INVITE sip:[hidden email] SIP/2.0.
> > > Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bK2e975bf5;rport.
> > > Max-Forwards: 70.
> > > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> > > To: <sip:[hidden email]>.
> > > Contact: <sip:15141234567@192.168.1.108>.
> > > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> > > <mailto:641cab3f73fa37a871818d1a70c4061b@192.168.1.108>.
> > > CSeq: 102 INVITE.
> > > User-Agent: Asterisk PBX 1.6.0.6.
> > > Date: Wed, 29 Apr 2009 15:38:18 GMT.
> > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> > > Supported: replaces, timer.
> > > Content-Type: application/sdp.
> > > Content-Length: 265.
> > > .
> > > v=0.
> > > o=root 1992389746 1992389746 IN IP4 192.168.1.108.
> > > s=Asterisk PBX 1.6.0.6.
> > > c=IN IP4 192.168.1.108.
> > > t=0 0.
> > > m=audio 11232 RTP/AVP 0 101.
> > > a=rtpmap:0 PCMU/8000.
> > > a=rtpmap:101 telephone-event/8000.
> > > a=fmtp:101 0-16.
> > > a=silenceSupp:off - - - -.
> > > a=ptime:20.
> > > a=sendrecv.
> > >
> > > #
> > > U 1.1.1.1:5060 -> 192.168.1.108:5060
> > > SIP/2.0 100 Giving a try.
> > > Via: SIP/2.0/UDP
> > >
> >
> 192.168.1.108:5060;branch=z9hG4bK2e975bf5;rport=5060;received=74.56.45.88.
> > > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> > > To: <sip:[hidden email]>.
> > > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> > > <mailto:641cab3f73fa37a871818d1a70c4061b@192.168.1.108>.
> > > CSeq: 102 INVITE.
> > > Server: OpenSIPS (1.4.4-notls (x86_64/linux)).
> > > Content-Length: 0.
> > > .
> > >
> > > #
> > > U 1.1.1.1:5060 -> 192.168.1.108:5060
> > > SIP/2.0 183 Session Progress.
> > > Via: SIP/2.0/UDP
> > >
> >
> 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060.
> > > To: <sip:[hidden email]>;tag=as664de2c2.
> > > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> > > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> > > <mailto:641cab3f73fa37a871818d1a70c4061b@192.168.1.108>.
> > > CSeq: 102 INVITE.
> > > Content-Type: application/sdp.
> > > Contact: <sip:15141234567@2.2.2.2:5060>.
> > > Content-Length: 241.
> > > Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
> > > User-Agent: Packetrino.
> > > Supported: replaces.
> > > Record-Route: <sip:2.2.2.2:5060;lr>.
> > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> > > .
> > > v=0.
> > > o=root 29378 29378 IN IP4 64.2.142.160.
> > > s=session.
> > > c=IN IP4 1.1.1.1.
> > > t=0 0.
> > > m=audio 52528 RTP/AVP 0 101.
> > > a=rtpmap:0 PCMU/8000.
> > > a=rtpmap:101 telephone-event/8000.
> > > a=fmtp:101 0-16.
> > > a=silenceSupp:off - - - -.
> > > a=ptime:20.
> > > a=sendrecv.
> > >
> > > #
> > > U 1.1.1.1:5060 -> 192.168.1.108:5060
> > > SIP/2.0 180 Ringing.
> > > Via: SIP/2.0/UDP
> > >
> >
> 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060.
> > > To: <sip:[hidden email]>;tag=as664de2c2.
> > > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> > > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> > > <mailto:641cab3f73fa37a871818d1a70c4061b@192.168.1.108>.
> > > CSeq: 102 INVITE.
> > > Contact: <sip:15141234567@2.2.2.2:5060>.
> > > Content-Length: 0.
> > > Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
> > > User-Agent: Packetrino.
> > > Supported: replaces.
> > > Record-Route: <sip:2.2.2.2:5060;lr>.
> > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> > > .
> > >
> > > #
> > > U 1.1.1.1:5060 -> 192.168.1.108:5060
> > > SIP/2.0 200 OK.
> > > Via: SIP/2.0/UDP
> > >
> >
> 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060.
> > > To: <sip:[hidden email]>;tag=as664de2c2.
> > > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> > > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> > > <mailto:641cab3f73fa37a871818d1a70c4061b@192.168.1.108>.
> > > CSeq: 102 INVITE.
> > > Content-Type: application/sdp.
> > > Contact: <sip:15141234567@2.2.2.2:5060>.
> > > Content-Length: 241.
> > > Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
> > > User-Agent: Packetrino.
> > > Supported: replaces.
> > > Record-Route: <sip:2.2.2.2:5060;lr>.
> > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> > > .
> > > v=0.
> > > o=root 29378 29379 IN IP4 64.2.142.160.
> > > s=session.
> > > c=IN IP4 1.1.1.1.
> > > t=0 0.
> > > m=audio 52528 RTP/AVP 0 101.
> > > a=rtpmap:0 PCMU/8000.
> > > a=rtpmap:101 telephone-event/8000.
> > > a=fmtp:101 0-16.
> > > a=silenceSupp:off - - - -.
> > > a=ptime:20.
> > > a=sendrecv.
> > >
> > > #
> > > U 192.168.1.108:5060 -> 2.2.2.2:5060
> > > ACK sip:15141234567@2.2.2.2:5060 SIP/2.0.
> > > Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bK04335252;rport.
> > > Route: <sip:2.2.2.2:5060;lr>,<sip:1.1.1.1;lr=on;nat=yes>.
> > > Max-Forwards: 70.
> > > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> > > To: <sip:[hidden email]>;tag=as664de2c2.
> > > Contact: <sip:15141234567@192.168.1.108>.
> > > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> > > <mailto:641cab3f73fa37a871818d1a70c4061b@192.168.1.108>.
> > > CSeq: 102 ACK.
> > > User-Agent: Asterisk PBX 1.6.0.6.
> > > Content-Length: 0.
> > > .
> > >
> > >
> > >
> ------------------------------------------------------------------------
> > >
> > > _______________________________________________
> > > Users mailing list
> > > [hidden email]
> > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> > >
> >
>


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Re: handling multiple proxy / Record-Route

Bogdan-Andrei Iancu
In reply to this post by Julien Chavanton
I think the only thing you can do is (if this path is fixed), to simply
ignore the Route headers and to do a static routing.

Regards,
Bogdan

Julien Chavanton wrote:

> I think I will try the option to use the "textops" module to enforce
> the correct order of Record-Route to validate this is my problem etc.
>  
>  
>
> ------------------------------------------------------------------------
> *From:* [hidden email] on behalf of Julien Chavanton
> *Sent:* Thu 30/04/2009 3:44 PM
> *To:* Bogdan-Andrei Iancu
> *Cc:* [hidden email]
> *Subject:* Re: [OpenSIPS-Users] handling multiple proxy / Record-Route
>
> thank you, this is a problem as I do not control this proxy (2.2.2.2),
> is there a suggested way of handling this problem ?
>  
> Maybe there is something esle wrong on my side cusaing this problem so
> I am including the SIP communication between the proxy this time
>  
>  
>  
> #
> U 1.1.1.1:5060 -> 2.2.2.2:5060
> INVITE sip:15148622633@2.2.2.2 SIP/2.0.
> Record-Route: <sip:1.1.1.1;lr>.
> Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bK09e6.36a0f975.0.
> Via: SIP/2.0/UDP
> 10.0.1.74:58366;received=10.0.1.74;branch=z9hG4bK-d87543-0f348609f47bda44-1--d87543-;rport=58366.
> Max-Forwards: 69.
> Contact: <sip:777@10.0.1.74:58366>.
> To: "15141234567"<sip:[hidden email]>.
> From: "777"<sip:[hidden email]>;tag=a030735d.
> Call-ID: 8116f933cc4ea03fMjYzN2Q1MGQ5Y2M1ZDc5Yzk4OTRjN2Y5YzEwYWMwMzc..
> CSeq: 1 INVITE.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
> SUBSCRIBE, INFO.
> Content-Type: application/sdp.
> User-Agent: eyeBeam release 1003s stamp 31159.
> Content-Length: 478.
> P-hint: Route[6]: mediaproxy .
> .
> v=0.
> o=- 8 2 IN IP4 10.0.1.74.
> s=CounterPath eyeBeam 1.5.
> c=IN IP4 1.1.1.1.
> t=0 0.
> m=audio 52550 RTP/AVP 0 8 18 101.
> a=alt:1 4 : LM6OZaAl 4x8r9qea 192.168.1.101 50006.
> a=alt:2 3 : 84SVypDj oi4PbxZ7 192.168.114.1 50006.
> a=alt:3 2 : L4wf6+MH s4gK5GAV 192.168.146.1 50006.
> a=alt:4 1 : cg2pbkCG WDFvj29+ 10.0.1.74 50006.
> a=fmtp:18 annexb=no.
> a=fmtp:101 0-15.
> a=rtpmap:101 telephone-event/8000.
> a=sendrecv.
> a=x-rtp-session-id:D56BCBC26473491FA111854E4C9F3575.
> a=direction:active.
> #
> U 2.2.2.2:5060 -> 1.1.1.1:5060
> SIP/2.0 100 Trying.
> Via: SIP/2.0/UDP
> 1.1.1.1:5060;branch=z9hG4bK09e6.36a0f975.0;received=1.1.1.1;rport=5060.
> Via: SIP/2.0/UDP
> 10.0.1.74:58366;received=10.0.1.74;branch=z9hG4bK-d87543-0f348609f47bda44-1--d87543-;rport=58366.
> To: "15141234567" <sip:[hidden email]>.
> From: "777" <sip:[hidden email]>;tag=a030735d.
> Call-ID: 8116f933cc4ea03fMjYzN2Q1MGQ5Y2M1ZDc5Yzk4OTRjN2Y5YzEwYWMwMzc..
> CSeq: 1 INVITE.
> Contact: <sip:15141234567@2.2.2.2>.
> Content-Length: 0.
> Record-Route: <sip:1.1.1.1;lr>.
> User-Agent: Packetrino.
> Supported: replaces.
> Record-Route: <sip:2.2.2.2:5060;lr>.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> .
>  
>
> ------------------------------------------------------------------------
> *From:* Bogdan-Andrei Iancu [mailto:[hidden email]]
> *Sent:* Thu 30/04/2009 3:44 PM
> *To:* Julien Chavanton
> *Cc:* [hidden email]
> *Subject:* Re: [OpenSIPS-Users] handling multiple proxy / Record-Route
>
> Hi Julian,
>
> Julien Chavanton wrote:
> >
> >
> > UA --> PROXY 1.1.1.1 --> PROXY 2.2.2.2 --> UA
> >
> > P1 --> P2
> > INVITE
> > Record-Route: <sip:1.1.1.1;lr=on;nat=yes>
> >
> > P2 --> P1
> > 100 Trying
> > Record-Route: <sip:1.1.1.1;lr=on;nat=yes>
> > Record-Route: <sip:2.2.2.2:5060;lr>
> >
> ^^^^^^^^^^^^
>
> This is not correct. The RR of P2 most me on top of RR of P1 - adding RR
> headers works as a stack.
>
> Regards,
> Bogdan
> >
> > Is there something wrong ? shouldn't proxy 2.2.2.2 add his
> > Record-Route on top of the existing Record-Route ?
> >
> > ------------------------------------------------------------------------
> > *From:* Bogdan-Andrei Iancu [mailto:[hidden email]]
> > *Sent:* Thu 30/04/2009 8:12 AM
> > *To:* Julien Chavanton
> > *Cc:* [hidden email]
> > *Subject:* Re: [OpenSIPS-Users] handling multiple proxy / Record-Route
> >
> > Hi Julien,
> >
> > I think Asterisk is doing the job properly. As you see the 200 OK has:
> >     Contact: <sip:15141234567@2.2.2.2:5060>.
> >     Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
> >   Record-Route: <sip:2.2.2.2:5060;lr>.
> >
> > So, Asterisk is generating the ACK with the Contact in RURI and the
> > Route set in the reverted order (correct loose routing).
> >     -> RURI: sip:15141234567@2.2.2.2:5060
> >            Destination: sip:2.2.2.2:5060;lr
> >      Route: sip:2.2.2.2:5060;lr + sip:1.1.1.1;lr=on;nat=yes
> >
> > I think the problem here is who and why adding the bottom RR in 200 OK
> > (why 2 of them ?)
> >
> > Regards,
> > Bogdan
> >
> > Julien Chavanton wrote:
> > >
> > > Hi,
> > >
> > > I have a situation whit multiple proxy where ACK is not sent as I
> > > would expect.
> > >
> > > if we look at the following "200 OK", I am expecting ACK to be sent to
> > > 1.1.1.1 but the "Asterisk PBX 1.6.0.6." is selecting 2.2.2.2 is this
> > > normal ?
> > >
> > > Do I have to handle Record-Route differently ?
> > >
> > >
> > >
> > >
> > >
> > > U 1.1.1.1:5060 -> 192.168.1.108:5060
> > > SIP/2.0 200 OK.
> > > Via: SIP/2.0/UDP
> > >
> >
> 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060.
> > > To: <sip:[hidden email]>;tag=as664de2c2.
> > > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> > > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> > > <mailto:641cab3f73fa37a871818d1a70c4061b@192.168.1.108>.
> > > CSeq: 102 INVITE.
> > > Content-Type: application/sdp.
> > > Contact: <sip:15141234567@2.2.2.2:5060>.
> > > Content-Length: 241.
> > > Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
> > > User-Agent: Packetrino.
> > > Supported: replaces.
> > > Record-Route: <sip:2.2.2.2:5060;lr>.
> > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > > ---------------------------------------------------------
> > >
> > > complete SIP signaling
> > >
> > > ---------------------------------------------------------
> > >
> > > #
> > > U 192.168.1.108:5060 -> 1.1.1.1:5060
> > > INVITE sip:[hidden email] SIP/2.0.
> > > Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bK2e975bf5;rport.
> > > Max-Forwards: 70.
> > > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> > > To: <sip:[hidden email]>.
> > > Contact: <sip:15141234567@192.168.1.108>.
> > > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> > > <mailto:641cab3f73fa37a871818d1a70c4061b@192.168.1.108>.
> > > CSeq: 102 INVITE.
> > > User-Agent: Asterisk PBX 1.6.0.6.
> > > Date: Wed, 29 Apr 2009 15:38:18 GMT.
> > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> > > Supported: replaces, timer.
> > > Content-Type: application/sdp.
> > > Content-Length: 265.
> > > .
> > > v=0.
> > > o=root 1992389746 1992389746 IN IP4 192.168.1.108.
> > > s=Asterisk PBX 1.6.0.6.
> > > c=IN IP4 192.168.1.108.
> > > t=0 0.
> > > m=audio 11232 RTP/AVP 0 101.
> > > a=rtpmap:0 PCMU/8000.
> > > a=rtpmap:101 telephone-event/8000.
> > > a=fmtp:101 0-16.
> > > a=silenceSupp:off - - - -.
> > > a=ptime:20.
> > > a=sendrecv.
> > >
> > > #
> > > U 1.1.1.1:5060 -> 192.168.1.108:5060
> > > SIP/2.0 100 Giving a try.
> > > Via: SIP/2.0/UDP
> > >
> >
> 192.168.1.108:5060;branch=z9hG4bK2e975bf5;rport=5060;received=74.56.45.88.
> > > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> > > To: <sip:[hidden email]>.
> > > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> > > <mailto:641cab3f73fa37a871818d1a70c4061b@192.168.1.108>.
> > > CSeq: 102 INVITE.
> > > Server: OpenSIPS (1.4.4-notls (x86_64/linux)).
> > > Content-Length: 0.
> > > .
> > >
> > > #
> > > U 1.1.1.1:5060 -> 192.168.1.108:5060
> > > SIP/2.0 183 Session Progress.
> > > Via: SIP/2.0/UDP
> > >
> >
> 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060.
> > > To: <sip:[hidden email]>;tag=as664de2c2.
> > > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> > > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> > > <mailto:641cab3f73fa37a871818d1a70c4061b@192.168.1.108>.
> > > CSeq: 102 INVITE.
> > > Content-Type: application/sdp.
> > > Contact: <sip:15141234567@2.2.2.2:5060>.
> > > Content-Length: 241.
> > > Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
> > > User-Agent: Packetrino.
> > > Supported: replaces.
> > > Record-Route: <sip:2.2.2.2:5060;lr>.
> > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> > > .
> > > v=0.
> > > o=root 29378 29378 IN IP4 64.2.142.160.
> > > s=session.
> > > c=IN IP4 1.1.1.1.
> > > t=0 0.
> > > m=audio 52528 RTP/AVP 0 101.
> > > a=rtpmap:0 PCMU/8000.
> > > a=rtpmap:101 telephone-event/8000.
> > > a=fmtp:101 0-16.
> > > a=silenceSupp:off - - - -.
> > > a=ptime:20.
> > > a=sendrecv.
> > >
> > > #
> > > U 1.1.1.1:5060 -> 192.168.1.108:5060
> > > SIP/2.0 180 Ringing.
> > > Via: SIP/2.0/UDP
> > >
> >
> 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060.
> > > To: <sip:[hidden email]>;tag=as664de2c2.
> > > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> > > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> > > <mailto:641cab3f73fa37a871818d1a70c4061b@192.168.1.108>.
> > > CSeq: 102 INVITE.
> > > Contact: <sip:15141234567@2.2.2.2:5060>.
> > > Content-Length: 0.
> > > Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
> > > User-Agent: Packetrino.
> > > Supported: replaces.
> > > Record-Route: <sip:2.2.2.2:5060;lr>.
> > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> > > .
> > >
> > > #
> > > U 1.1.1.1:5060 -> 192.168.1.108:5060
> > > SIP/2.0 200 OK.
> > > Via: SIP/2.0/UDP
> > >
> >
> 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060.
> > > To: <sip:[hidden email]>;tag=as664de2c2.
> > > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> > > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> > > <mailto:641cab3f73fa37a871818d1a70c4061b@192.168.1.108>.
> > > CSeq: 102 INVITE.
> > > Content-Type: application/sdp.
> > > Contact: <sip:15141234567@2.2.2.2:5060>.
> > > Content-Length: 241.
> > > Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
> > > User-Agent: Packetrino.
> > > Supported: replaces.
> > > Record-Route: <sip:2.2.2.2:5060;lr>.
> > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> > > .
> > > v=0.
> > > o=root 29378 29379 IN IP4 64.2.142.160.
> > > s=session.
> > > c=IN IP4 1.1.1.1.
> > > t=0 0.
> > > m=audio 52528 RTP/AVP 0 101.
> > > a=rtpmap:0 PCMU/8000.
> > > a=rtpmap:101 telephone-event/8000.
> > > a=fmtp:101 0-16.
> > > a=silenceSupp:off - - - -.
> > > a=ptime:20.
> > > a=sendrecv.
> > >
> > > #
> > > U 192.168.1.108:5060 -> 2.2.2.2:5060
> > > ACK sip:15141234567@2.2.2.2:5060 SIP/2.0.
> > > Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bK04335252;rport.
> > > Route: <sip:2.2.2.2:5060;lr>,<sip:1.1.1.1;lr=on;nat=yes>.
> > > Max-Forwards: 70.
> > > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> > > To: <sip:[hidden email]>;tag=as664de2c2.
> > > Contact: <sip:15141234567@192.168.1.108>.
> > > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> > > <mailto:641cab3f73fa37a871818d1a70c4061b@192.168.1.108>.
> > > CSeq: 102 ACK.
> > > User-Agent: Asterisk PBX 1.6.0.6.
> > > Content-Length: 0.
> > > .
> > >
> > >
> > >
> ------------------------------------------------------------------------
> > >
> > > _______________________________________________
> > > Users mailing list
> > > [hidden email]
> > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> > >
> >
>


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Re: handling multiple proxy / Record-Route

Julien Chavanton
Re: [OpenSIPS-Users] handling multiple proxy / Record-Route
Thanks ,a programmer of packetrino have requested a trace I am waiting for there answer.
 


From: Bogdan-Andrei Iancu [mailto:[hidden email]]
Sent: Sat 02/05/2009 8:17 AM
To: Julien Chavanton
Cc: [hidden email]
Subject: Re: [OpenSIPS-Users] handling multiple proxy / Record-Route

I think the only thing you can do is (if this path is fixed), to simply
ignore the Route headers and to do a static routing.

Regards,
Bogdan

Julien Chavanton wrote:


> I think I will try the option to use the "textops" module to enforce
> the correct order of Record-Route to validate this is my problem etc.


>
> ------------------------------------------------------------------------
> *From:* [hidden email] on behalf of Julien Chavanton
> *Sent:* Thu 30/04/2009 3:44 PM
> *To:* Bogdan-Andrei Iancu
> *Cc:* [hidden email]
> *Subject:* Re: [OpenSIPS-Users] handling multiple proxy / Record-Route
>
> thank you, this is a problem as I do not control this proxy (2.2.2.2),
> is there a suggested way of handling this problem ?

> Maybe there is something esle wrong on my side cusaing this problem so
> I am including the SIP communication between the proxy this time



> #
> U 1.1.1.1:5060 -> 2.2.2.2:5060
> INVITE sip:15148622633@2.2.2.2 SIP/2.0.
> Record-Route: <sip:1.1.1.1;lr>.
> Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bK09e6.36a0f975.0.
> Via: SIP/2.0/UDP
> 10.0.1.74:58366;received=10.0.1.74;branch=z9hG4bK-d87543-0f348609f47bda44-1--d87543-;rport=58366.
> Max-Forwards: 69.
> Contact: <sip:777@10.0.1.74:58366>.
> To: "15141234567"<sip:[hidden email]>.
> From: "777"<sip:[hidden email]>;tag=a030735d.
> Call-ID: 8116f933cc4ea03fMjYzN2Q1MGQ5Y2M1ZDc5Yzk4OTRjN2Y5YzEwYWMwMzc..
> CSeq: 1 INVITE.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
> SUBSCRIBE, INFO.
> Content-Type: application/sdp.
> User-Agent: eyeBeam release 1003s stamp 31159.
> Content-Length: 478.
> P-hint: Route[6]: mediaproxy .
> .
> v=0.
> o=- 8 2 IN IP4 10.0.1.74.
> s=CounterPath eyeBeam 1.5.
> c=IN IP4 1.1.1.1.
> t=0 0.
> m=audio 52550 RTP/AVP 0 8 18 101.
> a=alt:1 4 : LM6OZaAl 4x8r9qea 192.168.1.101 50006.
> a=alt:2 3 : 84SVypDj oi4PbxZ7 192.168.114.1 50006.
> a=alt:3 2 : L4wf6+MH s4gK5GAV 192.168.146.1 50006.
> a=alt:4 1 : cg2pbkCG WDFvj29+ 10.0.1.74 50006.
> a=fmtp:18 annexb=no.
> a=fmtp:101 0-15.
> a=rtpmap:101 telephone-event/8000.
> a=sendrecv.
> a=x-rtp-session-id:D56BCBC26473491FA111854E4C9F3575.
> a=direction:active.
> #
> U 2.2.2.2:5060 -> 1.1.1.1:5060
> SIP/2.0 100 Trying.
> Via: SIP/2.0/UDP
> 1.1.1.1:5060;branch=z9hG4bK09e6.36a0f975.0;received=1.1.1.1;rport=5060.
> Via: SIP/2.0/UDP
> 10.0.1.74:58366;received=10.0.1.74;branch=z9hG4bK-d87543-0f348609f47bda44-1--d87543-;rport=58366.
> To: "15141234567" <sip:[hidden email]>.
> From: "777" <sip:[hidden email]>;tag=a030735d.
> Call-ID: 8116f933cc4ea03fMjYzN2Q1MGQ5Y2M1ZDc5Yzk4OTRjN2Y5YzEwYWMwMzc..
> CSeq: 1 INVITE.
> Contact: <sip:15141234567@2.2.2.2>.
> Content-Length: 0.
> Record-Route: <sip:1.1.1.1;lr>.
> User-Agent: Packetrino.
> Supported: replaces.
> Record-Route: <sip:2.2.2.2:5060;lr>.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> .

>
> ------------------------------------------------------------------------
> *From:* Bogdan-Andrei Iancu [[hidden email]]
> *Sent:* Thu 30/04/2009 3:44 PM
> *To:* Julien Chavanton
> *Cc:* [hidden email]
> *Subject:* Re: [OpenSIPS-Users] handling multiple proxy / Record-Route
>
> Hi Julian,
>
> Julien Chavanton wrote:
> >
> >
> > UA --> PROXY 1.1.1.1 --> PROXY 2.2.2.2 --> UA
> >
> > P1 --> P2
> > INVITE
> > Record-Route: <sip:1.1.1.1;lr=on;nat=yes>
> >
> > P2 --> P1
> > 100 Trying
> > Record-Route: <sip:1.1.1.1;lr=on;nat=yes>
> > Record-Route: <sip:2.2.2.2:5060;lr>
> >
> ^^^^^^^^^^^^
>
> This is not correct. The RR of P2 most me on top of RR of P1 - adding RR
> headers works as a stack.
>
> Regards,
> Bogdan
> >
> > Is there something wrong ? shouldn't proxy 2.2.2.2 add his
> > Record-Route on top of the existing Record-Route ?
> >
> > ------------------------------------------------------------------------
> > *From:* Bogdan-Andrei Iancu [[hidden email]]
> > *Sent:* Thu 30/04/2009 8:12 AM
> > *To:* Julien Chavanton
> > *Cc:* [hidden email]
> > *Subject:* Re: [OpenSIPS-Users] handling multiple proxy / Record-Route
> >
> > Hi Julien,
> >
> > I think Asterisk is doing the job properly. As you see the 200 OK has:
> >     Contact: <sip:15141234567@2.2.2.2:5060>.
> >     Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
> >   Record-Route: <sip:2.2.2.2:5060;lr>.
> >
> > So, Asterisk is generating the ACK with the Contact in RURI and the
> > Route set in the reverted order (correct loose routing).
> >     -> RURI: sip:15141234567@2.2.2.2:5060
> >            Destination: sip:2.2.2.2:5060;lr
> >      Route: sip:2.2.2.2:5060;lr + sip:1.1.1.1;lr=on;nat=yes
> >
> > I think the problem here is who and why adding the bottom RR in 200 OK
> > (why 2 of them ?)
> >
> > Regards,
> > Bogdan
> >
> > Julien Chavanton wrote:
> > >
> > > Hi,
> > >
> > > I have a situation whit multiple proxy where ACK is not sent as I
> > > would expect.
> > >
> > > if we look at the following "200 OK", I am expecting ACK to be sent to
> > > 1.1.1.1 but the "Asterisk PBX 1.6.0.6." is selecting 2.2.2.2 is this
> > > normal ?
> > >
> > > Do I have to handle Record-Route differently ?
> > >
> > >
> > >
> > >
> > >
> > > U 1.1.1.1:5060 -> 192.168.1.108:5060
> > > SIP/2.0 200 OK.
> > > Via: SIP/2.0/UDP
> > >
> >
> 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060.
> > > To: <sip:[hidden email]>;tag=as664de2c2.
> > > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> > > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> > > <[hidden email]>.
> > > CSeq: 102 INVITE.
> > > Content-Type: application/sdp.
> > > Contact: <sip:15141234567@2.2.2.2:5060>.
> > > Content-Length: 241.
> > > Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
> > > User-Agent: Packetrino.
> > > Supported: replaces.
> > > Record-Route: <sip:2.2.2.2:5060;lr>.
> > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > > ---------------------------------------------------------
> > >
> > > complete SIP signaling
> > >
> > > ---------------------------------------------------------
> > >
> > > #
> > > U 192.168.1.108:5060 -> 1.1.1.1:5060
> > > INVITE sip:[hidden email] SIP/2.0.
> > > Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bK2e975bf5;rport.
> > > Max-Forwards: 70.
> > > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> > > To: <sip:[hidden email]>.
> > > Contact: <sip:15141234567@192.168.1.108>.
> > > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> > > <[hidden email]>.
> > > CSeq: 102 INVITE.
> > > User-Agent: Asterisk PBX 1.6.0.6.
> > > Date: Wed, 29 Apr 2009 15:38:18 GMT.
> > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> > > Supported: replaces, timer.
> > > Content-Type: application/sdp.
> > > Content-Length: 265.
> > > .
> > > v=0.
> > > o=root 1992389746 1992389746 IN IP4 192.168.1.108.
> > > s=Asterisk PBX 1.6.0.6.
> > > c=IN IP4 192.168.1.108.
> > > t=0 0.
> > > m=audio 11232 RTP/AVP 0 101.
> > > a=rtpmap:0 PCMU/8000.
> > > a=rtpmap:101 telephone-event/8000.
> > > a=fmtp:101 0-16.
> > > a=silenceSupp:off - - - -.
> > > a=ptime:20.
> > > a=sendrecv.
> > >
> > > #
> > > U 1.1.1.1:5060 -> 192.168.1.108:5060
> > > SIP/2.0 100 Giving a try.
> > > Via: SIP/2.0/UDP
> > >
> >
> 192.168.1.108:5060;branch=z9hG4bK2e975bf5;rport=5060;received=74.56.45.88.
> > > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> > > To: <sip:[hidden email]>.
> > > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> > > <[hidden email]>.
> > > CSeq: 102 INVITE.
> > > Server: OpenSIPS (1.4.4-notls (x86_64/linux)).
> > > Content-Length: 0.
> > > .
> > >
> > > #
> > > U 1.1.1.1:5060 -> 192.168.1.108:5060
> > > SIP/2.0 183 Session Progress.
> > > Via: SIP/2.0/UDP
> > >
> >
> 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060.
> > > To: <sip:[hidden email]>;tag=as664de2c2.
> > > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> > > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> > > <[hidden email]>.
> > > CSeq: 102 INVITE.
> > > Content-Type: application/sdp.
> > > Contact: <sip:15141234567@2.2.2.2:5060>.
> > > Content-Length: 241.
> > > Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
> > > User-Agent: Packetrino.
> > > Supported: replaces.
> > > Record-Route: <sip:2.2.2.2:5060;lr>.
> > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> > > .
> > > v=0.
> > > o=root 29378 29378 IN IP4 64.2.142.160.
> > > s=session.
> > > c=IN IP4 1.1.1.1.
> > > t=0 0.
> > > m=audio 52528 RTP/AVP 0 101.
> > > a=rtpmap:0 PCMU/8000.
> > > a=rtpmap:101 telephone-event/8000.
> > > a=fmtp:101 0-16.
> > > a=silenceSupp:off - - - -.
> > > a=ptime:20.
> > > a=sendrecv.
> > >
> > > #
> > > U 1.1.1.1:5060 -> 192.168.1.108:5060
> > > SIP/2.0 180 Ringing.
> > > Via: SIP/2.0/UDP
> > >
> >
> 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060.
> > > To: <sip:[hidden email]>;tag=as664de2c2.
> > > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> > > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> > > <[hidden email]>.
> > > CSeq: 102 INVITE.
> > > Contact: <sip:15141234567@2.2.2.2:5060>.
> > > Content-Length: 0.
> > > Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
> > > User-Agent: Packetrino.
> > > Supported: replaces.
> > > Record-Route: <sip:2.2.2.2:5060;lr>.
> > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> > > .
> > >
> > > #
> > > U 1.1.1.1:5060 -> 192.168.1.108:5060
> > > SIP/2.0 200 OK.
> > > Via: SIP/2.0/UDP
> > >
> >
> 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060.
> > > To: <sip:[hidden email]>;tag=as664de2c2.
> > > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> > > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> > > <[hidden email]>.
> > > CSeq: 102 INVITE.
> > > Content-Type: application/sdp.
> > > Contact: <sip:15141234567@2.2.2.2:5060>.
> > > Content-Length: 241.
> > > Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
> > > User-Agent: Packetrino.
> > > Supported: replaces.
> > > Record-Route: <sip:2.2.2.2:5060;lr>.
> > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> > > .
> > > v=0.
> > > o=root 29378 29379 IN IP4 64.2.142.160.
> > > s=session.
> > > c=IN IP4 1.1.1.1.
> > > t=0 0.
> > > m=audio 52528 RTP/AVP 0 101.
> > > a=rtpmap:0 PCMU/8000.
> > > a=rtpmap:101 telephone-event/8000.
> > > a=fmtp:101 0-16.
> > > a=silenceSupp:off - - - -.
> > > a=ptime:20.
> > > a=sendrecv.
> > >
> > > #
> > > U 192.168.1.108:5060 -> 2.2.2.2:5060
> > > ACK sip:15141234567@2.2.2.2:5060 SIP/2.0.
> > > Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bK04335252;rport.
> > > Route: <sip:2.2.2.2:5060;lr>,<sip:1.1.1.1;lr=on;nat=yes>.
> > > Max-Forwards: 70.
> > > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.
> > > To: <sip:[hidden email]>;tag=as664de2c2.
> > > Contact: <sip:15141234567@192.168.1.108>.
> > > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108
> > > <[hidden email]>.
> > > CSeq: 102 ACK.
> > > User-Agent: Asterisk PBX 1.6.0.6.
> > > Content-Length: 0.
> > > .
> > >
> > >
> > >
> ------------------------------------------------------------------------
> > >
> > > _______________________________________________
> > > Users mailing list
> > > [hidden email]
> > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> > >
> >
>


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