[asterisk-users] No reply to SIP OPTIONS - sip peers becoming randomly UNREACHABLE

[Asterisk]

Hi guys,

Am having a strange SIP problem in my call centre. The call centre has about 70 SIP agents (some of the are using SIP hard phones, other SIP softphones), and occasionally most of the SIP peers (hardphones and softphones) become UNREACHABLE and then after few second again REACHABLE. Some hardphones and softphones work perfectly normal during that period (even normally responding to OPTIONS message), but most of them get UNREACHABLE.

I don't have NAT - phones and Asterisk are in the same subnet, so nothing complicated really (regarding network configuration).

I'm currently suspecting my network to be the problem, but I would just like to confirm with you guys, if you have any similar experiences, what could be causing this?

Please, see bellow one of the sample SIP traces.

Regards,
Alex

Jan  1 11:17:42 VERBOSE[6046] logger.c: Reliably Transmitting (no NAT) to 165.11.1.41:5060:
OPTIONS sip:TestPhone1 at 165.11.1.41 SIP/2.0
Via: SIP/2.0/UDP 165.11.1.50:5060;branch=a4bG4bK4b7hf375;rport
From: "asterisk" <sip:asterisk at 165.11.1.50>;tag=as02e1afaa
To: <sip:TestPhone1 at 165.11.1.41>
Contact: <sip:asterisk at 165.11.1.50>
Call-ID: 3fa169320586bad01cd93bd87adf1c30 at 165.11.1.50
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 01 Jan 2010 11:17:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

Jan  1 11:17:45 VERBOSE[6046] logger.c: Retransmitting #1 (no NAT) to 165.11.1.41:5060:
OPTIONS sip:TestPhone1 at 165.11.1.41 SIP/2.0
Via: SIP/2.0/UDP 165.11.1.50:5060;branch=a4bG4bK4b7hf375;rport
From: "asterisk" <sip:asterisk at 165.11.1.50>;tag=as02e1afaa
To: <sip:TestPhone1 at 165.11.1.41>
Contact: <sip:asterisk at 165.11.1.50>
Call-ID: 3fa169320586bad01cd93bd87adf1c30 at 165.11.1.50
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 01 Jan 2010 11:17:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

Jan  1 11:17:46 NOTICE[6046] chan_sip.c: Peer 'TestPhone1' is now UNREACHABLE!  Last qualify: 14

Jan  1 11:17:56 VERBOSE[6046] logger.c: Reliably Transmitting (no NAT) to 165.11.1.41:5060:
OPTIONS sip:TestPhone1 at 165.11.1.41 SIP/2.0
Via: SIP/2.0/UDP 165.11.1.50:5060;branch=z2h16b637dKh2fd;rport
From: "asterisk" <sip:asterisk at 165.11.1.50>;tag=as796f6356
To: <sip:TestPhone1 at 165.11.1.41>
Contact: <sip:asterisk at 165.11.1.50>
Call-ID: 3367c4dc6cbdd57d67b0c5b53d5491e2 at 165.11.1.50
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 01 Jan 2010 11:17:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

Jan  1 11:17:56 VERBOSE[6046] logger.c: 
<-- SIP read from 165.11.1.41:5060: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 165.11.1.50:5060;branch=z2h16b637dKh2fd;rport
From: "asterisk" <sip:asterisk at 165.11.1.50>;tag=as796f6356
To: <sip:TestPhone1 at 165.11.1.41>;tag=5A4BF5F8-460290A9
CSeq: 102 OPTIONS
Call-ID: 3367c4dc6cbdd57d67b0c5b53d5491e2 at 165.11.1.50
Contact: <sip:TestPhone1 at 165.11.1.41>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.2.0.0047
Content-Length: 0
Jan  1 11:17:56 VERBOSE[6046] logger.c: --- (10 headers 0 lines) ---
Jan  1 11:17:56 NOTICE[6046] chan_sip.c: Peer 'TestPhone1' is now REACHABLE! (16ms / 10000ms)