Scenario#32 – SIP Calls drop after 75 minutes

I had an interesting case where SIP calls over a SIP trunk were dropping after like 75 minutes. The duration was not confirmed as sometimes it use to drop even before 75 minutes.

I looked into few CCSIP debugs (debug ccsip messages) and found that the ‘BYE’ message was actually coming from our end (Call manager/Gateway).

Platform information:

CCM: System version: 7.1.3.32900-4

CUBE-10#sh ver

Cisco IOS Software, 3800 Software (C3825-ADVIPSERVICESK9-M), Version 12.4(24)T4, RELEASE SOFTWARE (fc2)

Technical Support: http://www.cisco.com/techsupport
Copyright (c) 1986-2010 by Cisco Systems, Inc.
Compiled Fri 03-Sep-10 09:15 by prod_rel_team

ROM: System Bootstrap, Version 12.4(13r)T, RELEASE SOFTWARE (fc1)

CUBE-10 uptime is 27 weeks, 3 days, 44 minutes

voice-card 0
!
!
voice call send-alert
voice rtp send-recv
!
voice service voip
allow-connections sip to sip
fax protocol cisco
sip
options-ping 180
!
!
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711alaw

!

voice class sip-profiles 1
request INVITE sip-header Allow-Header modify “, UPDATE” “”

!

dial-peer voice 10 voip
description *** Outbound to GXXX – SIP Provider ***
translation-profile outgoing OUTBOUND
huntstop
destination-pattern 9T
voice-class codec 1
voice-class sip early-offer forced
voice-class sip profiles 1
session protocol sipv2
session target ipv4:10.0.222.69
dtmf-relay rtp-nte
fax-relay sg3-to-g3
no vad

!
!

gateway
timer receive-rtp 1200

!
sip-ua
retry invite 1
retry response 2
retry bye 2
retry cancel 1
retry options 1
timers trying 200
!

!

!

Debug Output:

Jul 11 12:17:54.579 BST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

BYE sip:079XXXXXXX@10.0.222.69:5060 SIP/2.0

Via: SIP/2.0/UDP 10.0.222.65:5060;branch=z9hG4bK100B17D
From: “anonymous” <sip:anonymous@10.0.222.65>;tag=B891B2E8-282
To: <sip:079XXXXXXX@10.0.222.69>;tag=3519367399-656588
Date: Mon, 11 Jul 2011 11:15:27 GMT
Call-ID: D687F817-AADB11E0-A725957A-CA524E0B@10.0.222.65
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Timestamp: 1310383074
CSeq: 128 BYE
Reason: Q.850;cause=16
Content-Length: 0

Jul 11 12:17:54.591 BST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.0.222.65:5060;branch=z9hG4bK100B17D
To: <sip:079XXXXXXX@10.0.222.69>;tag=3519367399-656588
From: “anonymous” <sip:anonymous@10.0.222.65>;tag=B891B2E8-282
Call-ID: D687F817-AADB11E0-A725957A-CA524E0B@10.0.222.65
CSeq: 128 BYE

Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH

Contact: <sip:079XXXXXXX@10.0.222.69:5060>

Content-Leng

= = = =

This is what I have done to fix the problem:

I added these commands under sip-profile:

voice class sip-profiles 1
request INVITE sip-header Allow-Header modify “UPDATE,” “”      < < <  This was already there
request REINVITE sip-header Allow-Header modify ” UPDATE,” “”
response 200 sip-header Allow-Header modify “UPDATE,” “”
response 180 sip-header Allow-Header modify “UPDATE,” “”

The next step was to increase the Service Parameter “SIP Session Expire Timer” which you can find under Service Parameters > Call manager:

That was it, it fixed the issue and customer confirmed having a call for more than 3 hours.

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4 thoughts on “Scenario#32 – SIP Calls drop after 75 minutes

  1. hi,

    If there are any NAT translations are there.. Make sure you have the inside Static NAT for the CME Router/Cisco Call Manager.

    If no NAT is there, need to check the SIP Expiry timer. It defines the maximum time that an INVITE message remains valid. If CME/CCM has not received an answer before this timer expires, CME/CCM terminates the call

  2. I Ashar. its me Amritesh (If you rememeber), good to see you researching on Advance topic…
    I got stuck with the same problem but it get disconnects after 60 mins aprox…
    I applied your command and waiting for result from the customer.. Did you also added the SIP profiles to the dial-peers???

  3. Hi Amrit, good to see you man…Yeah the profile was already there, I just added few commands under SIP profile.

    You try first with service parameter, if that doesn’t help then add SIP profile.

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