OAISYS SIP Integration

Transcription

1 OAISYS SIP Integration 08/26/2013 Americas Headquarters OAISYS 7965 South Priest Drive, Suite 105 Tempe, AZ USA (480)

2 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. OAISYS SIP Integration 1

3 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. OAISYS SIP Integration 2

4 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 OAISYS SIP Integration 3

5 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. OAISYS SIP Integration 4

6 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 5

7 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 OAISYS SIP Integration 6

8 2. Smart Control Board Tab view of HPX virtual board 3. View of license information window OAISYS SIP Integration 7

9 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 OAISYS SIP Integration 8

10 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 OAISYS SIP Integration 9

11 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. OAISYS SIP Integration 10

12 SIP CALL The following image shows how a SIP call appears in the OAISYS Management Studio. OAISYS SIP Integration 11

13 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 OAISYS SIP Integration 12

14 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. OAISYS SIP Integration 13

15 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 OAISYS SIP Integration 14

16 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. OAISYS SIP Integration 15

17 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. OAISYS SIP Integration 16

18 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: [ :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 OAISYS SIP Integration 17

19 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. OAISYS SIP Integration 18

20 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 , option 4! OAISYS SIP Integration 19