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Configuring a Unified Border Element with a Vitelity SIP Trunk w/ UCM 7

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Configuring a Unified Border Element with a Vitelity SIP Trunk w/ UCM 7

fiestas.cesar
How to configure a Cisco Unified Border Element with a Vitelity SIP Trunk to
Be used by Cisco Unified Communications Manager 7 as a way out to the PSTN.


Lately the term "Sip trunk" has been gaining popularity among UC engineers and
customers /clients. Although I feel pretty comfortable setting up sip trunks,
Debugging sip messages, etc in asterisk. But I didn?t feel the same
with Unified Communications Manager up until a few weeks ago when I started to
to learn and test with SIP in a Cisco Environment.
[Internet]---[cable modem]---[2811]---[3560]---[CallManager]?[EndPoint]
------[H323 Between Router and CallManager] ?[SIP or SCCP]

This test involves the following elements:
-Cisco Unified Communications Manager 7.0.1.11001-8
-Cisco 2811 Voice Gateway loaded with c2800nm-ipvoice_ivs-mz.124-22.T.bin
-A Sip Trunk provided by Vitelity [vitelity.com]
-A Broadband Connection to the internet provided by Road Runner.
-A Cisco 7961 IP Phone with Load File SIP41.8-4-1S (SIP)
-A Cisco 7941 IP Phone with Load File SIP41.8-4-1S (SIP)
-A Cisco 7920 IP Phone with Firmware 7920.4.0-03-02(SCCP)
-EyeBeam by Counterpath (SIP) used a third party sip endpoint (will consume 3 DLU?s)
Also keep in mind that I am not using a demarcation router or anything like that, this test was just conducted using a cable modem with broadband internet service only, QOS was implemented internally, hence my voice traffic going back to vitelity was ?as is? via internet so most likely my voice traffic can or will be affected by data bursts at some point, thus this may degrade the quality of the call. I have yet to test this set up with a T1 connection where bandwidth is guaranteed (up and down)


Codec used was g711ulaw
My last roundtrip to 64.2.142.30 was clocked at average 71ms
CLTLAB-2811#ping 64.2.142.30
Type escape sequence to abort.
Sending 5, 100-byte ICMP Echos to 64.2.142.30, timeout is 2 seconds:
!!!!!
Success rate is 100 percent (5/5), round-trip min/avg/max = 64/71/80 ms
The total hops to 64.2.142.30 was 19
My Internet speed was 3632kb/s down and 337kb/s up.
DTMF was tested and all work great
The voice stream was measured at 84kb during an active call.

Anyway I will now proceed to explain what it takes to configure a vitelity SIP trunk in a Cisco 2811 running an ivs IOS code so that it can be used by Unified Communications Manager as a way out to the PSTN.
-First and forth most you need to open an account and/or purchase a DID with vitelity [vitelity.com] then once vitelity have assigned you a DID, make sure the DID routing is set to SIP Server.
-Your vitelity username and password are your Vitelity Sip Trunk credentials.
-The vitelity proxy is sip4.vitelity.net or 64.2.142.30
-My assigned DID is 70490906XX
Step 1 ? H323 Gateway Configuration
Create and Configure an H323 Gateway in Unified Communications Manager.
Device Information---------------------------------------------------------------
Device Name* : Enter the IP address of the Router
Description: Any description for example CUBE
Device Pool: Select the appropriate Device Pool
Common Device Configuration : Select your preference
Call classification*: use system default
Media Resource Group List: Select the appropriate MRGL
Packet Capture Mode*: Select your preference or leave as None
Packet Capture Duration: 0
Location: Select the appropriate Location
AAR Group: N/A unless you have one that you want to use
Tunneled Protocol*: None
Use Trusted Relay Point*: Default
Signaling Port*: 1720
[Check] Media Termination Point Required
[Check]Retry Video Call As Audio
Leave unchecked Wait for Far End H.245 Terminal Capability Set
Leave unchecked Transmit UTF-8 for calling Party Name
Leave unchecked SRTP allowed
Multilevel Precedence and Preemption(MLPP) Information----------------------------------
Leave all settings as default
Call Routing Information ? Inbound Calls ---------------------------------------------------------
Significant Digits*: All
Calling Search Space: Select the appropriate Calling Search Space
AAR Calling Search Space: None Unless you have one
Prefix DN: Leave blank
[Checked] Redirecting Number IE Delivery ? Inbound
[Checked]Enable Inbound FastStart
Call Routing Information ? Outbound Calls------------------------------------------------------
Calling Party Selection*: Originator
Calling Party Presentation*:Default
Called Party IE number type unknown*: Cisco CallManager
Calling Party IE number type unknown*:Cisco CallManager
Called Numbering Plan*: Cisco Callmanager
Calling Numbering Plan*: Cisco CallManager
Caller ID DN: Blank
Calling Routing Information ?Outbound Calls-----------------------------------------------
Calling Party Selection*: Originator
Calling Party Presentation*:Default
Called Party IE number type unknown*: Cisco CallManager
Calling Party IE number type unknown*:Cisco CallManager
Called Numbering Plan*: Cisco CallManager
Calling Numbering Plan: Cisco CallManager
Caller ID DN: Blank
Unchecked Display IE Delivery
Unchecked Redirecting Number IE Delivery ? Outbound
Unchecked Enable Outbound FastStart codec for Outbound Faststart
Called Party Transformation CSS: None
Unchecked Use Device Pool Called Party Transformation CSS calling Party transformation CSS
Unchecked Use Device Pool Calling Party Transformation CSS
Incoming Calling Party Settings------------------------------------------------------------------
Leave all as default
Click Save


Step 2 ? Call Routing Route Configuration
Associate the newly created H323 gateway with a RG, RL, also for this test I am using NAMP:PreDot as selection for the Discard Digits field under the Route List Detail Configuration Page

Step 3
Create a local 10 Digits pattern to test the sip trunk

Step 4 ? Endpoint Access to Sip Trunk
Make sure your phones are in the proper partition as well that they have the appropriate Calling Search Space.
Step 5 ? CUBE configuration
I am assuming that you have the proper IOS image loaded in your Cisco Router, in my case I am using a Cisco Router 2811 and my IOS image is c2800nm-ipvoice_ivs-mz.124-22.T.bin.
Here is part of my configuration I will only post the most relevant part of my configuration in this document.
***********************************************************************************
This configuration enables the basic Cisco Unified Border Element functionality on a platform. This functionality terminates an incoming VoIP call and re-originates it with the use of an outbound VoIP dial-peer. The calls can be H.323 to SIP or SIP to SIP
!
voice service voip
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
!
!
This configuration creates the voice class for codec selection preference and assigns and identification tag.
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
!
!
Configure the incoming and outgoing dial-peers with the relevant protocol, DTMF type, and codec information
!
dial-peer voice 91 voip
destination-pattern .......... ---to match my inbound DID I could possibly make this more specific
voice-class codec 1
session target ipv4:10.10.100.254 --------------Unified Communications Manager IP address
incoming called-number .
dtmf-relay h245-alphanumeric
no vad
!
dial-peer voice 9 voip
destination-pattern .T
voice-class codec 1
session protocol sipv2
session target sip-server
session transport udp
incoming called-number 9T
dtmf-relay rtp-nte
!
!
sip-ua
credentials username freckles3 password 7 001D40041409002222 realm asterisk (this credentials are used to register ourselves with the inbound traffic sent from vitelity)
authentication username freckles3 password 7 120056150259232323 realm asterisk (this credentials are used to authenticate ourselves with vitelity proxy for outbound traffic purposes)
no remote-party-id
retry invite 2
retry register 2
registrar dns:sip4.vitelity.net expires 3600
sip-server dns:sip4.vitelity.net
host-registrar
!
Also when entering the credential and authentication parameters use ?password 0? followed by your vitelity password, ie
Authentication username yourusername password 0 your password realm asterisk
Step 6 - Translations
It may be necessary for you to create some translation patterns in order to match a specific extension, you can do this either in Unified Communications Manager or in the router, in this test I am using a voice translation rule at the VG.
Useful debugging commands
-debug ccsip messages
-show sip-ua register status
-debug voip ccapi inout (this command is used to debug h323)

-Here is a post of the ?show sip-ua register status
CLTLAB-2811#show sip-ua register status
Line peer expires(sec) registered
================================ ========== ============ ==========
freckles3 -1 47 yes
As you see above the freckles3 user is the vitality username that I was assigned by vitality.
-Here is the post of the ?debug ccsip messages? when receiving a call but not answering the ip phone
CLTLAB-2811#debug ccsip messages
SIP Call messages tracing is enabled
CLTLAB-2811#term mon
CLTLAB-2811#
*Nov 12 03:31:30.387: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:Your10DigitsDID@YourPublicIP:58380 SIP/2.0
Via: SIP/2.0/UDP 64.2.142.30:5060;branch=z9hG4bK43cdc920;rport
From: "IncomingPSTNNumber" <sip:IncomingPSTNNumber@64.2.142.30>;tag=as3470e94e
To: <sip:Your10DigitsDID@YourPublicIP:58380>
Contact: <sip:IncomingPSTNNumber@64.2.142.30>
Call-ID: 688ab6ce59302a8209770fd316c38c74@64.2.142.30
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 12 Nov 2008 03:30:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 508

v=0
o=root 4115 4115 IN IP4 64.2.142.30
s=session
c=IN IP4 64.2.142.30
t=0 0
m=audio 14746 RTP/AVP 0 3 18 4 8 111 5 10 7 110 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

*Nov 12 03:31:30.411: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Date: Wed, 12 Nov 2008 03:31:30 GMT
From: "IncomingPSTNNumber" <sip:IncomingPSTNNumber@64.2.142.30>;tag=as3470e94e
Allow-Events: telephone-event
Content-Length: 0
To: <sip:Your10DigitsDID@YourPublicIP:58380>
Call-ID: 688ab6ce59302a8209770fd316c38c74@64.2.142.30
Via: SIP/2.0/UDP 64.2.142.30:5060;branch=z9hG4bK43cdc920;rport
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE


*Nov 12 03:31:30.535: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 180 Ringing
Date: Wed, 12 Nov 2008 03:31:30 GMT
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
From: "IncomingPSTNNumber" <sip:IncomingPSTNNumber@64.2.142.30>;tag=as3470e94e
Allow-Events: telephone-event
Content-Length: 0
To: <sip:Your10DigitsDID@YourPublicIP:58380>;tag=A748C48-1E49
Contact: <sip:Your10DigitsDID@192.168.1.104:5060>
Call-ID: 688ab6ce59302a8209770fd316c38c74@64.2.142.30
Via: SIP/2.0/UDP 64.2.142.30:5060;branch=z9hG4bK43cdc920;rport
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE


*Nov 12 03:31:33.707: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
REGISTER sip:sip4.vitelity.net:5060 SIP/2.0
Date: Wed, 12 Nov 2008 03:31:33 GMT
From: <sip:freckles3@sip4.vitelity.net>;tag=A7498AC-2267
Timestamp: 1226460693
Content-Length: 0
User-Agent: Cisco-SIPGateway/IOS-12.x
To: <sip:freckles3@sip4.vitelity.net>
Contact: <sip:freckles3@192.168.1.104:5060>
Expires: 3600
Call-ID: 5F82FDBB-AF5211DD-8560B705-BEC6C08D
Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK173A22C1
CSeq: 1128 REGISTER
Max-Forwards: 70


*Nov 12 03:31:33.779: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK173A22C1;received=YourPublicIP
From: <sip:freckles3@sip4.vitelity.net>;tag=A7498AC-2267
To: <sip:freckles3@sip4.vitelity.net>
Call-ID: 5F82FDBB-AF5211DD-8560B705-BEC6C08D
CSeq: 1128 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:freckles3@64.2.142.30>
Content-Length: 0


*Nov 12 03:31:33.779: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK173A22C1;received=YourPublicIP
From: <sip:freckles3@sip4.vitelity.net>;tag=A7498AC-2267
To: <sip:freckles3@sip4.vitelity.net>;tag=as09adda30
Call-ID: 5F82FDBB-AF5211DD-8560B705-BEC6C08D
CSeq: 1128 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="72640296"
Content-Length: 0


*Nov 12 03:31:33.783: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
REGISTER sip:sip4.vitelity.net:5060 SIP/2.0
Date: Wed, 12 Nov 2008 03:31:33 GMT
Authorization: Digest username="freckles3",realm="asterisk",uri="sip:sip4.vitelity.net:5060",response="c17a2600c7133af25bf85c35ed6d39cf",nonce="72640296",algorithm=MD5
From: <sip:freckles3@sip4.vitelity.net>;tag=A7498AC-2267
Timestamp: 1226460693
Content-Length: 0
User-Agent: Cisco-SIPGateway/IOS-12.x
To: <sip:freckles3@sip4.vitelity.net>
Contact: <sip:freckles3@192.168.1.104:5060>
Expires: 3600
Call-ID: 5F82FDBB-AF5211DD-8560B705-BEC6C08D
Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK173BFE
CSeq: 1129 REGISTER
Max-Forwards: 70


*Nov 12 03:31:33.851: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK173BFE;received=YourPublicIP
From: <sip:freckles3@sip4.vitelity.net>;tag=A7498AC-2267
To: <sip:freckles3@sip4.vitelity.net>
Call-ID: 5F82FDBB-AF5211DD-8560B705-BEC6C08D
CSeq: 1129 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:freckles3@64.2.142.30>
Content-Length: 0


*Nov 12 03:31:33.855: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK173BFE;received=YourPublicIP
From: <sip:freckles3@sip4.vitelity.net>;tag=A7498AC-2267
To: <sip:freckles3@sip4.vitelity.net>;tag=as09adda30
Call-ID: 5F82FDBB-AF5211DD-8560B705-BEC6C08D
CSeq: 1129 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 60
Contact: <sip:freckles3@192.168.1.104:5060>;expires=60
Date: Wed, 12 Nov 2008 03:30:35 GMT
Content-Length: 0


*Nov 12 03:31:38.987: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
CANCEL sip:Your10DigitsDID@YourPublicIP:58380 SIP/2.0
Via: SIP/2.0/UDP 64.2.142.30:5060;branch=z9hG4bK43cdc920;rport
From: "IncomingPSTNNumber" <sip:IncomingPSTNNumber@64.2.142.30>;tag=as3470e94e
To: <sip:Your10DigitsDID@YourPublicIP:58380>
Call-ID: 688ab6ce59302a8209770fd316c38c74@64.2.142.30
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


*Nov 12 03:31:38.995: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Date: Wed, 12 Nov 2008 03:31:38 GMT
From: "IncomingPSTNNumber" <sip:IncomingPSTNNumber@64.2.142.30>;tag=as3470e94e
Content-Length: 0
To: <sip:Your10DigitsDID@YourPublicIP:58380>
Call-ID: 688ab6ce59302a8209770fd316c38c74@64.2.142.30
Via: SIP/2.0/UDP 64.2.142.30:5060;branch=z9hG4bK43cdc920;rport
CSeq: 102 CANCEL


*Nov 12 03:31:39.003: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 487 Request Cancelled
Reason: Q.850;cause=16
Date: Wed, 12 Nov 2008 03:31:38 GMT
From: "IncomingPSTNNumber" <sip:IncomingPSTNNumber@64.2.142.30>;tag=as3470e94e
Allow-Events: telephone-event
Content-Length: 0
To: <sip:Your10DigitsDID@YourPublicIP:58380>;tag=A748C48-1E49
Call-ID: 688ab6ce59302a8209770fd316c38c74@64.2.142.30
Via: SIP/2.0/UDP 64.2.142.30:5060;branch=z9hG4bK43cdc920;rport
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE


*Nov 12 03:31:39.063: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:Your10DigitsDID@YourPublicIP:58380 SIP/2.0
Via: SIP/2.0/UDP 64.2.142.30:5060;branch=z9hG4bK43cdc920;rport
From: "IncomingPSTNNumber" <sip:IncomingPSTNNumber@64.2.142.30>;tag=as3470e94e
To: <sip:Your10DigitsDID@YourPublicIP:58380>;tag=A748C48-1E49
Contact: <sip:IncomingPSTNNumber@64.2.142.30>
Call-ID: 688ab6ce59302a8209770fd316c38c74@64.2.142.30
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70

I will conduct further testing using this setup , such as conferencing and Video Testing.
This document was created by Cesar Fiestas