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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 [email protected].
OAISYS SIP Integration
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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.
OAISYS SIP 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.
OAISYS SIP Integration
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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!