Sip Client would disconnect from Cme every 18 second ?

hasanreza · ‎11-19-2010

Dear All ,

Hi

I have Cisco 2821 , on IOS Version 12.4(13r)T, RELEASE SOFTWARE (fc1) " c2800nm-advipservicesk9-mz[1].124-24.T.bin" and cme version 7.1 , running since the last 2 years without any problem , i had client getting connected remotely using pptp to cisco2821 (Virtual-Template) and would use Cisco communicator , but slow network connection the voice would have disturbance, breakup etc , , I then shifted to SIP protocol , the voice quality on the same network changed from just manageable to excellent , the voice on pptp was so clear on G711ulaw encoding , using sip client xlite ,as never before . The problem is that every 19-20 the call disconnects , I opened a tac case and worked about for month on the same but did not have any result as well.

I am attaching the running-config of the router with debugs collected in hope that someone would be able to help me

Debug collect

___________________________

Remote End (Windows Vista Home ) , 172.31.1.38/24 - PPTP(VPN) – Office CME 172.31.1.1/24

Remote End Extension (602) - Office Extension (500)

At Office end we have 4 MB connection and at Remote End we have 2Mb Connection

The below debugs have been collected, once the debug output is taken , logging has been cleared to avoid repetition ,

The Debug file has the following (debug-ccsip.txt)

1) debug ccsip calls

2) debug ccsip events

3) debug ccsip errors

4) debug ccapi in out

5) debug ccsip mess

Also please find attached file with (basic-cme.txt)

1) Show flash

2) Show ver

3) Show run

4) Show tech

I need to figure this thing out as i want to use a Cisco Linksys Spa 525G with a builtin VPN to use on the same network , and Black Berry Running Sip Client

Till Date I have successfully confired the following to work with CME

¨ Nokia Phone Running Nokia Call Connect for Cisco

¨ SPA 941 Phone

¨ SPA 2002

¨ IPhone

Guys Seriously need your help

Hasan reza

hasanreza@hotmail.com

David Smith · ‎11-22-2010

Hasan,

The issue here is that CCME sends 200 OK to the INVITE for the answer, but CCME never receives an ACK back from the phone. CCME sends 200 OK 7 times total, the initial 200 OK...followed by 6 retries. CCME never receives an ACK, so the call will fail.

You'll want to sniff either CCME or your PC to see if the 200 OK is getting there...and if so, is X-Lite responding with ACK.

032014: Nov 19 19:25:41.878 AST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Date: Fri, 19 Nov 2010 15:25:28 GMT
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
From: "602"<602>;tag=6a516a1f
Allow-Events: telephone-event
Supported: replaces
Supported: sdp-anat
Remote-Party-ID: "Reception" <500>;party=called;screen=no;privacy=off
Content-Length: 179
To: "500"<500>;tag=2E85097C-1025
Contact: <500>
Content-Disposition: session;handling=required
Content-Type: application/sdp
Call-ID: 3f0d3029b760e240YWVkZGJmNjFkNTEwN2U3Nzg2NGNiMjM5YzUzNzhmZGI.
Via: SIP/2.0/UDP 172.31.1.47:24040;branch=z9hG4bK-d87543-243a6f29e409c26d-1--d87543-;rport
CSeq: 1 INVITE
Server: Cisco-SIPGateway/IOS-12.x

v=0
o=CiscoSystemsSIP-GW-UserAgent 5879 2442 IN IP4 92.97.166.79
s=SIP Call
c=IN IP4 92.97.166.79
t=0 0
m=audio 18684 RTP/AVP 0
c=IN IP4 92.97.166.79
a=rtpmap:0 PCMU/8000

hasanreza · ‎11-22-2010

Dear David

First of all i cant express my happiness and gratitude , just for the fact that someone did answer , GOD BLESS .

Well shall do a packet capture at the client end using wireshark and then send the entire capture , i tried doing the below at the CCME but here is my problem .

ip traffic-export profile sniffer mode capture

bidirectional

!

since i get connected via pptp to router using a virtual-template , i enabled the traffic capture on the virtual template interface , but it does not capture any traffic what so ever , this is my output for the show ip int brief .

Acti-Systems#show ip int brief

Interface IP-Address OK? Method Status Prot

ocol

FastEthernet0/0 172.31.1.1 YES NVRAM up up

Service-Engine0/0 172.31.1.1 YES TFTP up up

FastEthernet0/1 unassigned YES NVRAM up up

ATM0/3/0 unassigned YES NVRAM administratively down down

ATM0/3/0.1 unassigned YES unset administratively down down

NVI0 172.31.1.1 YES unset up up

SSLVPN-VIF0 unassigned NO unset up up

Virtual-Access1 unassigned YES unset down down

Virtual-Template1 92.97.166.79 YES TFTP down down

Virtual-Access2 unassigned YES unset up up

Virtual-Access3 unassigned YES unset up up

Virtual-Access4 unassigned YES unset up up

Virtual-Access4.1 92.97.166.79 YES TFTP up up

Dialer1 92.97.166.79 YES IPCP up up

Dialer1200 unassigned YES NVRAM up up

Acti-Systems#show users

Line User Host(s) Idle Location

514 vty 0 admin idle 1d23h 172.31.1.33

515 vty 1 admin idle 1d23h 172.31.1.32

516 vty 2 admin idle 1d23h 172.31.1.34

517 vty 3 admin idle 03:53:50 172.31.1.38

518 vty 4 admin idle 02:06:02 172.31.1.39

519 vty 5 admin idle 01:57:06 172.31.1.39

520 vty 6 admin idle 00:01:12 94.59.227.216

*521 vty 7 admin idle 00:00:01 172.31.1.46

Interface User Mode Idle Peer Address

Vi3 PPPoE 00:00:00 195.229.244.44

Vi4.1 modest PPPoVPDN - 172.31.1.46 (This MY PPTP Session )

Acti-Systems#

Please just guide me as to how i need to go ahead ,also i have got some debug from the client end .

If I try the virtual Interface , here is the error.

Acti-Systems(config)#interface Virtual-Access4.1

% Please use virtual template to configure your virtual access

Acti-Systems(config)#

Well in either case shall collect the wireshark output at the system end and revert back

Regards

hasan Reza

David Smith · ‎11-22-2010

Hello Hasan,

I'm wondering if this is a simple routing issue...where the phone can reach the CCME (172.31.1.1), but CCME cannot reach the phone because of a routing issue.

Your voice register configs are bound to FE 0/0:

voice register global
mode cme
source-address 172.31.1.1 port 5060

interface FastEthernet0/0
ip address 172.31.1.1 255.255.255.0

But it appears that the source being used on CCME is 92.97.166.79 because of your default route:

ip route 0.0.0.0 0.0.0.0 Dialer1

Also possibly because of your NAT configs.

031900: Nov 19 19:21:39.774 AST: //3240/685F1C919571/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x49EF0158
State of The Call        : STATE_DEAD
TCP Sockets Used         : NO
Calling Number           : 602
Called Number            : 500
Source IP Address (Sig ): 92.97.166.79
Destn SIP Req Addr:Port : 172.31.1.47:24040
Destn SIP Resp Addr:Port : 172.31.1.47:24040
Destination Name         : 172.31.1.47

The SIP phone you're using has an IP of 172.31.1.47, same subnet as FE 0/0. If X-Lite is set up with a proxy/GW of 172.31.1.1...then this explains why the phone can get to CCME. But I bet you cannot reach 172.31.1.47 using a source of 92.97.166.79.

Can you run a few quick tests?

do the following:

ping 172.31.1.47 source 172.31.1.1

Does that work?

ping 172.31.1.47 source 92.97.166.79

Does that work?

If you can ping 172.31.1.47 sourcing off the FE interface IP of 172.31.1.1, but not 92.97.166.79 then you have a routing config issue.

I'm assuming if you did a debug of an outbound call from an SCCP phone to the X-lite...you would see a similar issue, no response to INVITE.

If you are able to ping sourcing off 172.31.1.1, try adding the below and retesting:

conf t

voice service voip

sip

bind all source-interface fastEthernet 0/0

Then retest.

If unsuccesful grab a debug ccsip message of both an outound and inbound call.

Thanks, -Dave