'Global IP Office Technical Tip 200'

IP Office Technical Tip

Tip no:

200

Release Date:

January 23, 2008

Region:

GLOBAL

IP Office Session Initiation Protocol (SIP) Configuration Primer

There are many Internet Telephony Service Providers (ITSP) offering SIP services and

the programming and operation of SIP can vary between them. As a new protocol for

deployment, mis-configuration is a leading cause of IP Office SIP issues reported to

Avaya Support. IP Office SIP in release 4.0.10 and greater can be programmed to use

options found in several Internet Engineering Task Force (IETF) SIP related Request For

Comment (RFC) documents. With this document as reference the user can adjust the IP

Office SIP configuration to match their respective ITSP's SIP messaging format.

Individuals using this document must have an understanding of SIP protocols and

standards found in the IETF RFC's. Where applicable throughout this document the

IETF RFC related to the SIP format discussed or displayed is listed. This serves to

validate IP Office operation per the specified IETF RFC and allow further review of the

subject matter presented. A basic understanding and knowledge of IP Office SIP

configuration can be found in Global IP Office Technical Tip 188 (Tech Tip 188).

NOTE 1 - Ensure your SIP ITSP is listed in the "ITSP Interoperability" table found in IP

Office Technical Bulletin 80 - General Availability of IP Office 4.0 Software and IP Office

500 page 48. Updates of this list may be found at

http://marketingtools.avaya.com/knowledgebase/.

NOTE 2 - Throughout this document the IP Office Monitor is used to view the SIP

message format after the IP Office SIP configuration changes. Monitor is not an IP Office

application supported for field use. The operation, function and display of Monitor can

change without notice or documentation to the field. The ITSP equipment may have a

capability similar to Monitor, inquire and use this ability during deployment if possible.

SIP connections can also be monitored using commercial or freeware packet capture

software.

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Forward

The following information, captures and displays are demonstrated using IP Office

release 4.0.10. The diagram on the next page represents a typical network configuration

connecting IP Office A and the ITSP's SIP Equipment.

Throughout this document all example calls are initiated from IP Office A extension

33302 (IP Office > User > SIP tab = 17328533302). All calls from the originating

extension will be dialed to (called number) number 1-732-852-2203 located within, or

available through, the ITSP network.

The IP Office Monitor SIP tab is set per the screen shot below and connect to IP Office

A (IP Address = 192.168.43.1)

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IP Office SIP Tel INVITE

As a starting reference point the IP Office SIP configuration to the ISTP SIP service is

configured per Tech Tip 188. With Monitor connected to IP Office A, a call is initiated

(dialed) from extension 33302 to 8 (designating the SIP Line in the short codes and

ARS) and number 17328522203. After the call is completed and disconnected, pause or

"file > save as" the Monitor to review the SIP information. The below represents the SIP

INVITE sent by IP Office A, extension 33302 (identified within the User > SIP tab as

number 17328522203) to dialed number 17328522203 on remote IP Office B. Search

for and identify the first "INVITE" originating from IP Office A.

Display / Saved from Monitor

---------------------------------------------------------------------------------------------------------------------------------------------

SIP Tx: UDP 192.168.43.1:5060 -> 192.168.42.242:5060

INVITE Tel:+17328522203 SIP/2.0

Via: SIP/2.0/UDP

192.168.43.1:5060;rport;branch=z9hG4bKfe625964dd867f9a89f19fafcc65d335

From: 17328533302 <sip:17328533302@example.com>;tag=a083c01064cbeccf

To: Tel:+17328522203

Call-ID: c1de74d5f65d53f7779166eb1a428bee@192.168.43.1

CSeq: 1962617718 INVITE

Contact: 17328533302 <sip:17328533302@192.168.43.1:5060;transport=udp>

Max-Forwards: 70

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE

Content-Type: application/sdp

Content-Length: 299

v=0

o=UserA 310281170 981016607 IN IP4 192.168.43.1

s=Session SDP

c=IN IP4 192.168.43.1

t=0 0

m=audio 49152 RTP/AVP 18 4 8 0 101

a=rtpmap:18 G729/8000

a=rtpmap:4 G723/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=fmtp:18 annexb = no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

-------------------------------------------------------------------------------------------------------------------------------------------

SIP Tel INVITE Review

SIP Tx: - This line identifies SIP Transmit (Tx) information from the IP Office the Monitor

is connected to. Received SIP information will be displayed in the trace beginning with

"SIP Rx:"

1) The INVITE request. It is "Tel" formatted and contains only the dialed number with the

SIP version Identifier. The called number (17328522203) is prefixed with a "+ " (plus

sign) denoting a Global E.164 number format per IETF RFC 3966. Note the IP Office

default to the SIP Tel format.

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2) The "From" field displays the defined SIP > User tab information, in this case number

17328522203, in addition to "17328522203@example.com". The User number and the

fully qualified domain name as entered into the SIP Line tab "ITSP Domain Name" entry.

The "From" number can be designated as an E.164 number if desired by placing the " + "

in front of the number in the User > SIP tab field in the call originating system. See "User

> SIP Tab" section on next page.

3) The "To" field is "Tel" formatted, displaying only the dialed number (17328522203)

prefixed with a "+ " (plus sign) denoting a Global E.164 number format (IETF RFC 3966).

4) Contact information

The User > SIP tab data is displayed (1732853302) by selecting "Use User Data"

in the SIP URI "Local URI" definition.

The IP address of the device sending the INVITE 192.168.43.1

The port the INVITE sent from 5060 set in the SIP Line "Network

Configuration"

The port type (TCP or UDP) udp set in the SIP Line "Network Configuration"

5) The SDP session being setup.

6) The type of SDP session being setup denoted by the "m=audio" (IETF RFC 4566).

The port being negotiated for the media stream 49152

The type of application/media RTP standard AVP (Audio, Video, Protocol).

7) A list of the media attributes denoted by the "a=" entries (IETF RFC 2833).

Available codec's for audio negotiation (the codec setting in the SIP Line

"Network Configuration" set for "Automatic").

"telephone-event/8000" denotes the RTP payload type and the default sampling

rate (8000).

8) "0-15" denotes the DTMF events supported within the "telephone-event" payload

(DTMF digits 0 to 15). Instead of sending actual DTMF tones, representations

of the DTMF tones are sent (what key pressed and for how long) and the

receiving site is expected to reproduce the DTMF tones locally (RFC 2833).

NOTE 3 - The SIP "Tel" URI described above is a valid SIP format per IETF RFC 3966.

It is possible some ITSP's may not support the Tel format at this time. If a different SIP

URI format is desired, changing the Tel URI format is the responsibility of the first

network proxy receiving the INVITE (the ITSP provider equipment) per RFC 3966.

User > SIP Tab - Calling Number E.164 designation

By setting Local URI, Contact, Display Name to `Use User Data' in the SIP URI

programming each User now displays a tab for a SIP "User Data" entry. These entries

can be individually programmed and are then be used to populate the SIP URI fields. By

default this field is populated with the "Name" entry of the User. The example below

displays the addition of the global E.164 number designator " + " (plus sign) to the

calling number (IETF RFC 3966).

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Configured WITHOUT the " + " (plus sign) E.164 number designator

Configured WITH the " + " (plus sign) E.164 number designator

After configuring the User > SIP tab local number with the " + " sign save the

configuration. Connect / reconnect Monitor to IP Office A, a test call is repeated. After

successfully completing the call and disconnecting, pause, or "file > save as" the Monitor

to review the SIP information. The below represents the SIP INVITE sent by IP Office

A, extension 33302 (identified within the User > SIP tab as number +17328533302) to

dialed number 17328522203. Search for and identify the "INVITE" originating from IP

Office A.

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Displayed / Saved from Monitor

----------------------------------------------------------------------------------------------------------------------------------------------

SIP Tx: UDP 192.168.43.1:5060 -> 192.168.42.242:5060

INVITE Tel:+17328522203 SIP/2.0

Via:

SIP/2.0/UDP

192.168.43.1:5060;rport;branch=z9hG4bKfe625964dd867f9a89f19fafcc65d335

From: +17328533302 <sip:+17328533302@example.com>;tag=a083c01064cbeccf

To: Tel:+17328522203

Call-ID: c1de74d5f65d53f7779166eb1a428bee@192.168.43.1

CSeq: 1962617718 INVITE

Contact: +17328533302 <sip:+17328533302@192.168.43.1:5060;transport=udp>

Max-Forwards: 70

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE

Content-Type: application/sdp

Content-Length: 299

-------------------------------------------------------------------------------------------------------------------

NOTE 4 - from this point forward only the Monitor fields changed or affected will be

displayed for review.

Note that the "From" and "Contact" fields of the call originator now include the Global

E.164 number designation. (Items #1 and #2).

IP Office SIP INVITE - no Domain Name

Removing the entry in the SIP Line "ITSP Domain Name" field will remove the Domain

Name from the INVITE and will automatically insert the IP address of the calling unit in

this location in the transmitted INVITE.

Domain Name Removed

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Display / Saved from Monitor

----------------------------------------------------------------------------------------------------------------------------------------------

SIP Tx: UDP 192.168.43.1:5060 -> 192.168.42.242:5060

INVITE Tel:+17328522203 SIP/2.0

Via:

SIP/2.0/UDP

192.168.43.1:5060;rport;branch=z9hG4bKfe625964dd867f9a89f19fafcc65d335

From: +17328533302 <sip:+17328533302@192.168.43.1>;tag=a083c01064cbeccf

To: Tel:+17328522203

Call-ID: c1de74d5f65d53f7779166eb1a428bee@192.168.43.1

CSeq: 1962617718 INVITE

Contact: +17328533302 <sip:+17328533302@192.168.43.1:5060;transport=udp>

Max-Forwards: 70

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE

Content-Type: application/sdp

Content-Length: 299

1) The "From" field now displays the SIP > User tab information and the IP Address of

the transmitting IP Office "+17328533302 @192.168.43.1". The domain name is

replaced by the IP address. This can be compared to the two previous Monitor displays

were the IP Office was configured with the domain name entry earlier in this document.

IP Office SIP INVITE (non-Tel format)

As previously reviewed, when configured per Tech Tip 188 the IP Office will default to

the SIP "Tel" format, a valid format per the SIP standards (RFC 3966). It is possible that

the ITSP currently does not support this format or requests a different SIP format

described in RFC 3261. To change the IP Office SIP format to a non-Tel format add the

below text to the SIP Line ARS Short Code.

Non-Tel format - SIP Line ARS Short Code

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Note the addition of the text after the Telephone Number "N" entry

"@192.168.42.242;user=phone". Add this text as listed inserting the ITSP's IP address

of their SIP equipment and save the IP Office configuration.

Non-Tel format - SIP Line ARS Short Code

With the above text added to the SIP Line ARS Short Code a call is initiated from

extension 33302 to number 17328522203. After the call is completed and disconnected,

pause or "file > save as" the Monitor to review the SIP information.

Display / Saved from Monitor

-------------------------------------------------------------------------------------------------------------------------------------------

SIP Tx: UDP 192.168.43.1:5060 -> 192.168.42.242:5060

INVITE sip:17328522203@192.168.42.242;user=phone SIP/2.0

Via:

SIP/2.0/UDP

192.168.43.1:5060;rport;branch=z9hG4bKa083c01064cbeccf170c59d70bb9a58e

From: 17328533302 <sip:17328533302@example.com>;tag=6e4d1c7bb5cc7a9c

To: <sip:17328522203@192.168.42.242;user=phone>

Call-ID: 1d852fc75e04499646a8b960e4ffbee2@192.168.43.1

CSeq: 1265249445 INVITE

Contact: 17328533302 <sip:17328533302@192.168.43.1:5060;transport=udp>

Max-Forwards: 70

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE

Content-Type: application/sdp

Content-Length: 301

----------------------------------------------------------------------------------------------------------------------------------------------

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The "INVITE" and the "To:" fields (1 and 2 above) have been changed from the SIP Tel

format. These fields are now formatted similar to the SIP URI formats described in RFC

3261 section 19.1.3. Additional changes can be made in the SIP Line ARS Short Code

to adjust the IP Office SIP INVITE to match that expected by the SIP ITSP per RFC

3261. Coordinate these changes with the SIP ITSP.

Examples:

"@192.168.42.242"

"@example.com"

IP Office SIP INVITE (no Tel format) Global E.164 numbering

When changing the IP Office SIP INVITE to the non-Tel format note that the E.164

number designator "+" (plus sign) is not automatically appended to the outgoing (called)

number. If required to use the non-Tel SIP format and the E.164 designator by the ITSP

this can be accommodated. Modify the IP Office SIP Line ARS Short Code by inserting

the designator at the beginning of the Telephone Number "N" as seen below.

+N"@192.168.42.242;user=phone"

Non-Tel SIP format Global E.164 numbering

With the above " + " designator added to the SIP Line ARS Short Code a call is initiated

from extension 33302 to number 17328522203. After the call is completed and

disconnected, pause or "file > save as" the Monitor to review the SIP information.

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Display / Saved from Monitor

--------------------------------------------------------------------------------------------------------------------------------------------

SIP Tx: UDP 192.168.43.1:5060 -> 192.168.42.242:5060

INVITE sip:0017328522203@192.168.42.242;user=phone SIP/2.0

Via: SIP/2.0/UDP

192.168.43.1:5060;rport;branch=z9hG4bK35175e7b81bd07d91694a8b06af07a6b

From: 17328533302 <sip:17328533302@example.com>;tag=de12aa85559051d7

To: <sip:0017328522203@192.168.42.242;user=phone>

Call-ID: 398855d9b9dfcd268ab5285cefef05d4@192.168.43.1

CSeq: 859474496 INVITE

Contact: 17328533302 <sip:17328533302@192.168.43.1:5060;transport=udp>

Max-Forwards: 70

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE

Content-Type: application/sdp

Content-Length: 301

---------------------------------------------------------------------------------------------------------------------------------------------

Note in the above trace that the called number located in the "INVITE" (1) and the "To:"

(2) fields have been modified. In each field / number the "+" has been replaced with the

equivalent "00" (zero, zero) - 00 being the prefix for international calls (RFC 3966).

Issued by:

Avaya GSS Tier 4 Support

Contact details:-

EMEA/APAC

NA/CALA

Tel: +44 1707 392200

Tel: +1 732 852 1955

Fax: +44 (0) 1707 376933

Fax: +1 732 852 1943

Email: gsstier4@avaya.com

Email: IPOUST4ENG@Avaya.com

Internet: http://www.avaya.com

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