Overview
Sip normalization is a method used to modify sip messages sent from the call manager out the sip trunk. Before call manager 8.5 if we had to modify the sip messages the same had to be done by using a CUBE and appliying sip profiles on dial peers.
It is normally required in a scenario when you are integrated with a third party sip server and they have specific requirements in terms of information coming from the cisco side.
Topology
Cisco call manager (8.5 and above) ---> sip trunk -----> third party sip server
Normalization Scripts
- Sip normalization is a C script which enables us to change the various headers fields of a SIP messgae such as invite , from to, 181 etc.
- We create a recusrive function call, and enter the queries in a if else format.
- Following are the steps to apply a normalization scrip on calls:
- On the call manager admin page go to device
- Navigate to device settings and select sip mormalization script under it.
- Click on add new. Name the scipt something according to your naming conventions.
Below is a example of a default script where in we need to change the from feild in the SIP invite message and add "user=phone" tag.
Default Invite
INVITE sip:6233284618@0.0.0.0:5060 SIP/2.0
Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK2938b58562
From: <sip:5555@0.0.0.0>;tag=241~f88ceed7-4ccc-49aa-ad02-4534616cc344-29414802
To: <sip:6233284618@0.0.0.0>
Date: Wed, 07 Mar 2012 02:14:07 GMT
Call-ID: 36ac8700-f561c46f-1a-500a0ac@0.0.0.0
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Cisco-Guid: 0917276416-0000065536-0000000037-0083927212
Session-Expires: 1800
P-Asserted-Identity: <sip:5555@0.0.0.0>
Remote-Party-ID: <sip:5555@0.0.0.0>;party=calling;screen=yes;privacy=off
Contact: <sip:5555@0.0.0.0:5060>
Max-Forwards: 70
Content-Length: 0
Modified Invite
INVITE sip:4690717@0.0.0.0;user=phone SIP/2.0
Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK380671373
From: 7024343344 <sip:7024343344@0.0.0.0;user=phone>;tag=00404d0102030a0a0a36-4950fca5
To: <sip:4690717@0.0.0.0;user=phone>
Call-ID: 1996806959@0.0.0.0
CSeq: 2 INVITE
Contact: <sip:7024343344@0.0.0.0;user=phone>
Authorization: Digest username="476119065202", realm="BroadWorks", nonce="BroadWorksXgt37bop3Td0105nBW", uri="sip:4690717@0.0.0.0;user=phone", response="4a8491694347451bd28dd2b5c35bc1a8", algorithm=MD5, cnonce="702c5d07", qop=auth, nc=00000001
Max-Forwards: 70
User-Agent: HST-3000/6.0.0
Unsupported: 100rel
Supported: resource-priority
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 133
As observed above we need to add a tag "user=phone" in the from message.
Normalization Script
M = {}
function M.outbound_INVITE(msg)
msg:addHeaderUriParameter("From", "user", "phone")
end
return M
This script is a recursive C function call. We created a function M.outbound_INVITE(msg) which is for outbound calls and will affect the invite message.
It will modify the from field to the desired results.
- After adding the script save the same.
- Go to the SIP trunk configured for out bound calls.
- Choose this script on the SIP trunk and then save it.
Important Links
Link for SIP normalization scripts:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/sip_tn/8_5_1/10-sip_transparency.html
Hope this helpful.
