OAISYS SIP Integration

08/26/2013

Americas Headquarters

OAISYS

7965 South Priest Drive, Suite 105

Tempe, AZ 85284

USA

www.oaisys.com

(480) 496-9040

OAISYS SIP Integration

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OVERVIEW

OAISYS introduced the ability to record calls that originate on a SIP trunk with version

6.1; OAISYS is currently on version 7.3. The Session Initiation Protocol (SIP) is a

signaling protocol used for controlling multimedia communication sessions such as

voice and video calls over Internet Protocol (IP) networks. This document provides

information on configuring the OAISYS Solution for SIP recording using SIP Info

Mapping or Matching Logic.

NOTE: OAISYS version 6.1 introduced SIP recording but did not include SMDR/CTI

Integration.

SIP INFO MAPPING AND MATCHING LOGIC

The Uniform Resource Identifier (URI) is a string of characters used to identify a name

or web resource. This identification enables interaction with representations of the web

resource over a network using specific protocols. The SIP URI is essentially a data

stream with defined protocols and syntaxes containing information which can be used to

help locate a call record when that information is mapped properly.

For call recording purposes, a SIP device refers to per call not per device (SIP can have

multiple active calls). The OAISYS solution integrates directly with SIP devices (trunk or

station) to record calls by capturing call data from the SIP device.

NOTE: Dynamic licensing is not applicable in this configuration.

Version 6.2 of the Recording Server Software introduces recording of SIP trunks with

Matching Logic. SMDR Matching Logic can be used with Allworx, Avaya IPOffice, Mitel

3300, Mitel 5000, Toshiba CIX, or ShoreTel to provide extension information and

account codes to the OAISYS recording server.

Matching Logic is not 100% accurate, but provides a close match to the criteria entered.

For example: if two calls took place at 10:23:35, lasting 30 seconds to the same outside

phone number, OAISYS Matching Logic could not make a match. If a call cannot be

distinctly matched, no extra information will be attached to any call. Whereas, recording

TDM Trunks with CTI is accurate all the time.

Recording SIP Trunks with Matching Logic differs from recording traditional T1 or PRI

trunks with CTI integration.

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SMDR Matching Logic on PBXs supporting multiple state transitions

ALL EXTENSIONS involved with the call will be attached to the call moments

after it is complete

NOTE: The Mitel 3300 and Toshiba CIX support multiple state transitions

On other PBXs

THE LAST EXTENSION involved with the call will be attached to the call shortly

after the call is complete

When recording on TDM trunks with CTI

ALL EXTENSIONS involved with a call are attached to the call record

The criteria that can be used for searching records and establishing permissions differ

between SIP Trunk with Matching Logic and the TDM Trunk with CTI.

See the comparison chart on the following page.

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Feature TDM Trunk with CTI SIP Trunk with

Matching Logic

Station Information Only after the call is complete

Account Code Only after the call is complete

Start Date & Time

Call Duration

Call Direction

Manual Start/Stop Recording

Caller ID

DNIS

ACD Agent

ACD Group

Extra Call Information Only after the call is complete

After Call Actions

Live Call Monitoring No extension info on live calls

Screen Recording Option

Desktop Client Application

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REQUIREMENTS

OAISYS Software Version 6.1 or later

o Please reference the RTP Configuration Guide for configuration steps

One call on a SIP trunk at one time

o One voice port required per call on a SIP trunk

OAISYS supports G.711 or G.729a

NOTE: Silence Suppression is not supported

o Calls using G.711 and G.729 streams simultaneously cannot be recorded

(this is a rare configuration).

Network Switch with Port Mirroring

AudioCodes USB Dongle and HPX License

o One monitor license per call

AudioCodes driver 5.7 required.

o Download from this location:

ftp://ftp.oaisys.com/pub/downloads/3rdparty/Ai-Logix/5.7/

SUPPORTED PBXS

OAISYS supports recording SIP Device Recording for the following PBXs:

Allworx

Avaya IPOffice

Mitel 3300

Mitel 5000

Toshiba CIX

ShoreTel

For PBXs not listed, please contact OAISYS Sales Engineering at SE@OAISYS.com.

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SIP TRUNK INTEGRATION DIAGRAM

EXPECTATIONS

The information available to the OAISYS solution when recording the SIP Trunk:

Start Date and Time

Call Duration

Call Direction

ANI/DNIS (if provided by the service provider)

This information can be used to search for calls and can be used to enable specific

permissions.

NOTE: IC calls or Peer-to-Peer calls are not recorded when using SIP Trunk

Integration.

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CONFIGURATION

The following information describes how to apply the AudioCodes license files and

configure the OAISYS solution to record audio on SIP devices.

1. Open AudioCode Smart Control through the control panel

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2. Smart Control Board Tab view of HPX virtual board

3. View of license information window

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4. Next, enable UDP port 5060 for SIP, to do this:

a. Open AudioCodes Smart View the board will indicate “CLOSED”

b. Open the board

c. This shows the board in “OPEN” state

5. Open the Signaling Protocol window

a. Enable UDP port 5060 for SIP

6. Open OAISYS Management Studio

a. From the Admin Tab, navigate to IP Endpoints SIP Devices Click on

the Plus sign to add new SIP Devices

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b. This will open the following pop-up window The SIP Device Type will

display SIP Trunk or SIP Station depending on the configuration

c. Enter a description

d. Enter the IP Address of the SIP Provider OR the IP Address of the Edge

Device (such as the router’s internal address)

e. Enter the SIP port number (default value is 5060)

f. Select Auto Generate

g. SIP to/from digits **use this only if recording SIP Trunks on a Mitel 5000**

h. The newly added SIP device information will appear as follows

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i. Add VoIP ports and select the adapter

j. Configure the port

NOTE: Late Binding checkbox enables a pool of recording ports. This is

the OAISYS recommended configuration option for SIP devices.

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SIP CALL

The following image shows how a SIP call appears in the OAISYS Management

Studio.

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SETUP SIP INFO MAPPING

This portion of the document covers the basic setup of an OAISYS Recording Server to

use SIP Mapping to correlate the SIP URI with the traditional telephony parameters to

attach SIP data to the call record. This data can be an important source of information

utilized to find a call.

Navigate to IP Endpoints SIP Info Mapping Click the Plus sign to set up a new SIP

map

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1. Name the SIP map in the Description field

2. In the Call Direction drop-down menu, choose the appropriate option

a. Ignore Call Direction

b. In To Device = Inbound

c. In To Device = Outbound

NOTE: If Ignore Call Direction is selected and the Call Direction data is not

reliably received, the OAISYS system will ignore the call direction and place data

into the fields as designated by the administrator in the Call In To Device and

Call Out From Device fields. It is important to recognize this limitation and select

the appropriate drop-down option carefully to avoid confusion.

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3. Map the SIP fields

NOTE: Not all fields need to be mapped – only the desired fields to be placed

into the call record

a. Select the appropriate SIP fields to map for the Call In To Device options

b. Select the appropriate SIP fields to map for the Call Out From Device

options

c. Save the changes

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4. Navigate to SIP Devices Highlight the appropriate SIP Device Click the Edit

button

a. Select the generated SIP Info map from the drop-down menu click

Save

The SIP URI information has been successfully mapped to the corresponding OAISYS

data field for the selected SIP Device. Repeat the SIP Info Mapping steps for any

remaining SIP Devices.

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SETUP MATCHING LOGIC

This portion of the document covers the basic setup of an OAISYS Recording Server

that has already been configured to record SIP trunks. This assumes the server is

already recording audio on the SIP channels, and it is now time to setup the Matching

Logic to get extension information on those calls.

1. To configure, associate SMDR Service with the PBX type (a Mitel 3300 Matching

Logic .DEF file is selected in the screen shot below), verify selection of the .DEF

file that has “Matching Logic” in the title for your PBX selection.

2. Expand Recording Manager select Recording Manager Status.

This section is to verify that if the system uses SIP Trunk only (no other recording

method), CTI must be disabled by choosing None for PBX integration by

extension.

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3. Select PBX Integration by Matching Logic to SMDR link.

a. All Mitel 3300 and CTX systems typically support Device State Transitions

(multiple SMDR per call) so check this box. This ensures there is only one

SMDR event per call (last known extension on the call).

b. The Mitel 5000 does not support Device State Transitions.

Make a few test calls to ensure the extension is bound to the call recording.

Once a call is complete, SMDR is seen from the PBX and place it into an event

queue. Approximately 5 minutes later, the system will run a database query to

determine if any calls match the criteria based on the SMDR event to match to

the call.

If a match is found, another query is run to add the information to the call.

If a match is not found initially, you will see:

[88204 07:34:49.7] [INFO]Fuzzy match failed for SMDR call data 2527 in

FuzzyMatchCallQueue 2; reason = No matches found” in the TRM events

o The system will run another attempt after 30 minutes; this is

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additional time allotted for the call to complete and be entered into

the database.

The query is run three (3) times: 5 minutes, 30 minutes, and 4 hours.

In some cases, the default hard-coded values in the timer settings need to be

changed. As the timing with every system is unique, some settings can be

refined to improve match quality.

Below are some example settings we have found are a good match:

These parameters are to set windows for the duration and start time of the

call and helps adjust for any latency or for system times that may not be

quite in synch.

HKLM\Software\Computer Telephony Solutions\Recording Manager

DWORD: Voice4NetDurationWindow(20)seconds

DWORD: Voice4NetStartTimeWindow(120)seconds

In the example above, it allows for matches within plus or minus 20

seconds of the known duration and plus or minus 120 seconds of the

known start time. So if a call was recorded and the known duration of the

recording is 5 minutes 25 seconds then potentially a match could occur

with an SMDR event that noted duration between 5 minutes 5 seconds

and 5 minutes 45 seconds.

The same is true for the start time window. If the known recording start

time was 2:00pm, then potentially, a match could occur with an SMDR

event that notes a start time between 1:58pm and 2:02pm.

These are the fuzzymatchqueue lookup timers (in seconds post call

completion):

In this example, the first match attempt would be 75 seconds after call

completion.

If no match, then a second attempt is run 15 seconds later.

If there still isn’t a match, thena final attempt is run 69 seconds later. The

initial queue (Queue0) should not be set to less than 60 seconds.

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HKLM\Software\Computer Telephony Solutions\Recording

Manager\FuzzyMatching

DWORD: Queue0DurationSeconds (75)

DWORD: Queue1DurationSeconds (15)

DWORD: Queue2DurationSeconds(69)

For further information or assistance, please contact Technical Support at 888-496-9040, option 4!