[asterisk-users] SIP OPTIONS response from the peer is ignored - peer becomes UNREACHABLE

[Asterisk]

Hi Steve,

Yes, that's true. It seems that Asterisk gets it with great delay. For instance:

Asterisk says:
==============

Mar 25 11:26:54 VERBOSE[1385] logger.c: Reliably Transmitting (no NAT) to 172.11.11.2:5060:
OPTIONS sip:MyTestPhone at 172.11.11.2 SIP/2.0
Via: SIP/2.0/UDP 172.11.0.201:5060;branch=z9hG4bK41333ebd;rport
From: "asterisk" <sip:asterisk at 172.11.0.201>;tag=as5e2d8165
To: <sip:MyTestPhone at 172.11.11.2>
Contact: <sip:asterisk at 172.11.0.201>
Call-ID: 5bf37c2d2d25a9346cc272d71c3a86e2 at 172.11.0.201
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 25 Mar 2010 10:26:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Mar 25 11:26:58 VERBOSE[1385] logger.c: Retransmitting #1 (no NAT) to 172.11.11.2:5060:
OPTIONS sip:MyTestPhone at 172.11.11.2 SIP/2.0
Via: SIP/2.0/UDP 172.11.0.201:5060;branch=z9hG4bK41333ebd;rport
From: "asterisk" <sip:asterisk at 172.11.0.201>;tag=as5e2d8165
To: <sip:MyTestPhone at 172.11.11.2>
Contact: <sip:asterisk at 172.11.0.201>
Call-ID: 5bf37c2d2d25a9346cc272d71c3a86e2 at 172.11.0.201
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 25 Mar 2010 10:26:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Mar 25 11:26:58 NOTICE[1385] chan_sip.c: Peer 'MyTestPhone' is now UNREACHABLE!  Last qualify: 52
Mar 25 11:26:58 VERBOSE[1385] logger.c: Destroying call '5bf37c2d2d25a9346cc272d71c3a86e2 at 172.11.0.201'

=== WEIRD: Asterisk logs two responses reposnses then (last one at 11:27:02 eventhough last OPTIONS request was sent by Asterisk at 11:26:58):

Mar 25 11:26:58 VERBOSE[1385] logger.c:
<-- SIP read from 172.11.11.2:5060: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.11.0.201:5060;branch=z9hG4bK41333ebd;rport
From: "asterisk" <sip:asterisk at 172.11.0.201>;tag=as5e2d8165
To: <sip:MyTestPhone at 172.11.11.2>;tag=CD9ADC40-8CECD6C3
CSeq: 102 OPTIONS
Call-ID: 5bf37c2d2d25a9346cc272d71c3a86e2 at 172.11.0.201
Contact: <sip:MyTestPhone at 172.11.11.2>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.2.0.0047
Content-Length: 0


Mar 25 11:26:58 VERBOSE[1385] logger.c: --- (10 headers 0 lines) ---
Mar 25 11:27:02 VERBOSE[1385] logger.c: 
<-- SIP read from 172.11.11.2:5060: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.11.0.201:5060;branch=z9hG4bK41333ebd;rport
From: "asterisk" <sip:asterisk at 172.11.0.201>;tag=as5e2d8165
To: <sip:MyTestPhone at 172.11.11.2>;tag=CD9ADC40-8CECD6C3
CSeq: 102 OPTIONS
Call-ID: 5bf37c2d2d25a9346cc272d71c3a86e2 at 172.11.0.201
Contact: <sip:MyTestPhone at 172.11.11.2>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.2.0.0047
Content-Length: 0


But Wireshark for the same conversation says:
=============================================

>> Sent by Asterisk at 11:26:54.047483000
-----------------------------------------
OPTIONS sip:MyTestPhone at 172.11.11.2 SIP/2.0
Via: SIP/2.0/UDP 172.11.0.201:5060;branch=z9hG4bK41333ebd;rport
From: "asterisk" <sip:asterisk at 172.11.0.201>;tag=as5e2d8165
To: <sip:MyTestPhone at 172.11.11.2>
Contact: <sip:asterisk at 172.11.0.201>
Call-ID: 5bf37c2d2d25a9346cc272d71c3a86e2 at 172.11.0.201
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 25 Mar 2010 10:26:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

<< Received from the phone at 11:26:54.097936000
------------------------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.11.0.201:5060;branch=z9hG4bK41333ebd;rport
From: "asterisk" <sip:asterisk at 172.11.0.201>;tag=as5e2d8165
To: <sip:MyTestPhone at 172.11.11.2>;tag=CD9ADC40-8CECD6C3
CSeq: 102 OPTIONS
Call-ID: 5bf37c2d2d25a9346cc272d71c3a86e2 at 172.11.0.201
Contact: <sip:MyTestPhone at 172.11.11.2>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.2.0.0047
Content-Length: 0

>> Sent by Asterisk at 11:26:58.486339000
-----------------------------------------
OPTIONS sip:MyTestPhone at 172.11.11.2 SIP/2.0
Via: SIP/2.0/UDP 172.11.0.201:5060;branch=z9hG4bK41333ebd;rport
From: "asterisk" <sip:asterisk at 172.11.0.201>;tag=as5e2d8165
To: <sip:MyTestPhone at 172.11.11.2>
Contact: <sip:asterisk at 172.11.0.201>
Call-ID: 5bf37c2d2d25a9346cc272d71c3a86e2 at 172.11.0.201
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 25 Mar 2010 10:26:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

<< Received from the phone at 11:26:58.524907000
------------------------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.11.0.201:5060;branch=z9hG4bK41333ebd;rport
From: "asterisk" <sip:asterisk at 172.11.0.201>;tag=as5e2d8165
To: <sip:MyTestPhone at 172.11.11.2>;tag=CD9ADC40-8CECD6C3
CSeq: 102 OPTIONS
Call-ID: 5bf37c2d2d25a9346cc272d71c3a86e2 at 172.11.0.201
Contact: <sip:MyTestPhone at 172.11.11.2>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.2.0.0047
Content-Length: 0


Hmmmmm, what could be causing such delay between Wireshark getting the data and Asterisk logging it?

Regards,
Alex

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steve Howes
Sent: Thursday, March 25, 2010 11:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP OPTIONS response from the peer is ignored - peer becomes UNREACHABLE

On 25 Mar 2010, at 10:18, Asterisk wrote:
> How is it possible that the peer becames UNREACHABLE eventhough Wireshark logged its proper response? 

Wireshark received it, doesn't mean Asterisk did. what does a sip debug in Asterisk show?

S
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