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200 OK with SDP not being sent to CCM

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200 OK with SDP not being sent to CCM
Answer
8/5/08 8:11 PM
My customer (Verizon) is testing CVP with CUBE for SIP trunking to the PSTN and is having the following problem.

Please see attached capture and CVP/CUBE configuration.

IP Addresses:

192.168.98.1 Private Side of Cube

192.168.98.10 Call Manager Server

192.168.98.30 CVP

192.168.98.40 Voice Browser

166.38.98.57 Public Side of Cube

166.34.84.77 Session Border Controller

Overview of problem:

When the terminating SIP phone places the PSTN caller on hold Call Manager sends out an INVITE without SDP which gets sent to the proxy and responded to with a 200 OK with SDP. The 200 OK with SDP never gets returned to Call Manager and therefore Call Manager never sends an ACK to the proxy which causes the call to be disconnected by the proxy.

The 200 OK is sent to CVP but CVP does not forward it to Call Manager only to the originating caller

Thanks,

Charles Demaret

CCIE#8287 R&S/Security

Cisco Systems Inc.

Cell: 972.249.6370

E-Mail: cdemaret@cisco.com
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Re: 200 OK with SDP not being sent to CCM
Answer
8/5/08 8:14 PM as a reply to Paul Antinori.
Hi Charles,

The problem is we are not seeing an ACK for this message that CVP is sending to CUBE, which is at time 22.731 in your pcap file. Could you grab the CUBE logs with full sip debugging turned on.

Thanks,

Paul

SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.98.1:5060;branch=z9hG4bK966270C
To: +17407774000 <sip:+17407774000@192.168.98.30:5060>;tag=dsc991678a
From: <sip:44441183@192.168.98.1;transport=udp>;tag=3ECABE34-1AE3
Call-ID: 12173635722011086@192.168.98.30
CSeq: 102 INVITE
Content-Length: 287
Cisco-Guid: 889384489-1558516189-3006718213-2961317982
Contact: <sip:+17407774000@192.168.98.30:5060;transport=udp>
Date: Tue, 29 Jul 2008 20:39:19 GMT
Server: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <sip:+17407774000@192.168.98.1>;party=called;screen=no;privacy=off
Supported: replaces
Content-Type: application/sdp
Session-Expires: 1800;refresher=uac

v=0
o=CiscoSystemsSIP-GW-UserAgent 8194 404 IN IP4 192.168.98.1
s=SIP Call
c=IN IP4 192.168.98.1
t=0 0
m=audio 16460 RTP/AVP 18 101
c=IN IP4 192.168.98.1
a=rtpmap:18 G729/8000
a=fmtp:18
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
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