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The VoIP tab is part of the System settings page, and contains settings for recording Voice over IP (VoIP) traffic.

 

VoIP network: Select on which network ports the recorder must listen for VoIP traffic.

- All: The recorder listens for VoIP traffic on all detected network ports, up to 8 maximum.

- On the "Phone" network socket: The recorder listens for VoIP traffic on the second network port (marked as "Phone" on default hardware), and any additional ports (up to 8 maximum), but excluding the first port (marked "PC" or "LAN" on default hardware).

- [ All + Internal ]: The recorder will listen for VoIP traffic on all detected network ports, and it addition, it will listen for VoIP traffic inside the recorder (the loopback virtual interface). Normally this option should not be chosen.

- None: The recorder does not listen for VoIP traffic on any device.

Check for faults: When enabled, the ethernet connection to the VoIP network will be checked (cable present or not, dropped packets), and included in the system health state.

Restart recording component: When a time is filled in, the system will restart the recording component if no calls are busy at that moment, each day. If a call is busy, it is rechecked a number of times, and if there are still active calls, then the recording component is not restarted. The connector to other recorders is restarted at the same time. Use the format HH:MM. Example: 02:30

VoIP recording module: The Default option can handle all the supported protocols. The Snip option can handle more load on multi-core systems, but supports only the SIP protocol. The Snip option lacks support for much of the special options, but it does support the following:

- in the System settings VoIP: SIP port, Local/Remote IP addresses, Local telephone numbers, Dial code action, Split long recordings

- in the Recording settings: Audio file encoding, Addresses to record, SIP Filter, Recording/delete/silence/store on demand, Record inc./outg./local calls, Call monitoring

- under Tools: Encryption of call content

Database link: This setting is normally set to Internal. This setting determines if the VoIP recording engine communicates with the database through an internal link, or through the web server (external).

PF_RING: Enables an advanced kernel driver, that reduces packet drop when under high load. It requires Debian 7 or higher to operate.

ERSPAN without header: Enable this option if you need to capture ERSPAN data that lacks the normal header.

IP address of PBX: The IP address of the VoIP telephone system. This usually not needs to be filled in. It is currently only used in combination with the next setting.

Do not record SIP calls that start at the PBX: When both the traffic to and from the PBX is seen by Apresa, this will result in double recordings in some situations. To prevent this, this option can be enabled.

SIP port: Comma-separated list of SIP ports (UDP/TCP).

For the Default module: If nothing valid is specified, all ports are checked for SIP data.

For the Optimized module: If nothing is specified, only port 5060 is checked for SIP data.

SIP INVITE determines IP addresses: If enabled, the IP addresses of the SIP invite, and not of the RTP, will be used to determine the direction of the call (incoming/outgoing).

Local IP Addresses: This setting is used to determine whether an IP address corresponds to a local phone (extension). It is also used for detecting of the direction of a call (incoming, outgoing, or internal). A singular IP address or an IP address range can be specified. To specify multiple IP addresses or ranges, separate them by a comma. An IP range must be specified in the CIDR notation. For example: 192.168.0.0/24 means that the first 24 bits are fixed, and the last 8 bits may vary, which means that, in this example, all IP addresses that start with 192.168.0. would be considered local.

Remote IP Addresses: In a VoIP call, there are two IP addresses that communicate. If one of them is in the Remote IP Address range, the other is consider local. The format is the same as for Local IP Addresses.

Local telephone numbers: This is a comma separated list of telephone numbers that are local. This setting is useful if it cannot be determined whether a call party is a local entity, based on IP address alone.

Instead of specifying telephone numbers one by one, it is possible to specify a range of numbers. For example to specify all numbers in the range from 3914100 to 3914199, specify: 3914100 - 3914199. The required format for number ranges is: First number space - space last number. The first and last number must have the same number of digits.

Alternatively, it is also possible use the * star sign at the front or at the end (not in the middle), to match with any number of characters at the front or end of the telephone number. For example, +33* will match any phone number that starts with +33, and another example, *1020 will match any phone number that ends on 1020.

MAC address as Local ID: When this option is enabled, the MAC address will be used as the identifier for local telephones. A name can be attached to the identifier.

Name as Local/Remote ID: In the VoIP protocols SIP and SCCP, a name is sometimes communicated. This option specifies when Apresa must use this name as identifier.

Use SIP name of remote party as: The SIP name is by default used to fill in the name of the contact belonging to the detected SIP ID. Alternatively and in special situations, the remote SIP name can be used as the Local ID, or to fill in the Connected field.

Usage of SIP ID as identification: SIP IDs are usually of the form number@host, for example 152@192.168.0.43. The default option specifies that only the part before the @ (usually the number) will be used for telephone identification (user access).

This can be useful in a multi-tenant environment. For example, the local number 140@1.2.3.4 can belong to tenant 1, while 140@4.5.6.7 can be long to tenant 2.

This option does not influence whether or not the full SIP ID is displayed in the main call listing.

Use P-Asserted-Identity as identifier (Caller): If enabled, the P-Asserted-Identity SIP header field will be used, instead of the From field, to identify the calling party.

Use P-Asserted-Identity as identifier (Receiver): If enabled, the P-Asserted-Identity SIP header field will be used, instead of the To field, to identify the receiving party.

If the P-Asserted-Identity field contains only an IP address, or a similar unusable identifier is detected, then the field will not be used, even if the setting is enabled.

Use Request-URI as identifier (Receiver): If enabled, the Request-URI (on the first line of the SIP INVITE) will be used, instead of the To field, to identify the receiving party. If the Request-URI does not contain a phone number, then the To field is used instead, even if this setting is enabled.

Use Remote-Party-ID as identifier (Caller) / (Receiver): If enabled, the Remote-Party-ID SIP header field will be used (if it contains something suitable) to identity the calling and/or receiving party.

SIP data field: Input a field or parameter that must be extracted from the SIP data, and select in which field the extracted data must be stored in the database. Or select the option SIP Filter if the data must not be stored but used for matching with the SIP Filter.

To extract the value of a SIP header field, enter the SIP header name without trailing colon. For example: User-Agent

To extract a parameter value in a SIP header field, input the SIP header name followed by a colon and the parameter name. For example: From:x-refci (this extracts the x-refci parameter from the From header field, which is the Cisco Call ID for active recording).

Tenant assignment field: Input a (custom) SIP field that must be extracted from the SIP data. If this data matches with what is configured in the settings of a tenant, the recording will be assigned to that tenant.

Use SIP to signal a recording must be stored: If store on demand is enabled, then the specified SIP field value can be used to signal that the recording must be stored. This SIP field value is expected to be in the first INVITE.

From secondary call legs, only use the Receiver ID: If enabled, the first call leg will determine the ID of the caller, while the last call leg will determine the ID of the receiver.

Split recording when telephone number changes: If enabled, recordings are split when the telephone number of one of the call participants changes during a call. It is applied in the following circumstances: During a special SIP call transfer, during a SIP identity change,  during a Cisco SCCP telephone number change, and during an change of remote ID of a Xpert call. If the option is disabled, the original telephone number is preserved, and the whole call is stored is one recording.

SIP REFER: When enabled, detects a special type of SIP transfer that uses REFER messages. Without this option, the caller ID would not be detected correctly.

Start recording when connected: When enabled, for the supported protocols, no recording will be made of the call setup period. The recording will exclude the time during which the phone was ringing. The recording starts when the connection is established.

Dialing prefix for outgoing calls: The digit or digits that must be dialed first when making an external call. Only set this option if remote numbers in outgoing calls contain an extra digit (or digits) at the start compared to remote numbers in incoming calls. The dialing prefix will be removed from the number in the call listing, and the call will be marked as outgoing.

Phone number prefixes to remove: This can be used to remove unwanted text from the front of the telephone number (for example the + sign). Multiple prefixes can be specified separated with a comma. This applies only to the SIP and the Cisco SCCP protocol.

Phone number suffixes to remove: This can be used to remove unwanted text from the end of the telephone number. The star (*) sign can be used at the end of the suffix to match any number of characters. It can be used to remove everything from the telephone number after and including a certain character sequence. Multiple suffices can be specified separated with a comma. This applies only to the SIP and the Cisco SCCP protocol. For a SIP ID of user@host, the suffix is applied only to the user part.

Detect phone number between parentheses: If the detected ID of the caller or receiver contains an ID between parentheses (round brackets), then the ID between the parentheses is used.

Audio stream can belong to multiple calls: When there are two SIP sessions, but only one audio stream, both SIP sessions will recorded if this option is enabled. Actions such as filtering can be performed on them separately. When the two SIP sessions are two legs of the same call, this option can lead to calls being recorded twice. Alternatively, if this option is disabled (which is the default currently), only one of the two sessions would be recorded.

Recording on demand is never applied to other phones in the SIP filter: Normally, a call is recorded on demand if one or both parties are in the recording on demand list. When this option is enabled, however, if one party is not in the recording on demand list, but it is in the SIP filter, then it is not recorded on demand, even when the other party is in the recording on demand list.

Maximum silence on demand time: If not empty, silence on demand is automatically stopped after the specified number of seconds of silence (with an accuracy of 5 seconds). When silence on demand is stopped, audio is again recorded.

Unpreferred SIP IDs: When there are two SIP sessions, and only one audio stream, preference will not be given to those SIP sessions whose telephone numbers (or IDs) are mentioned here (comma separated). For Lync/S4B, if a phone number is communicated in the Referred-By SIP field, this will replace an unpreferred SIP ID.

Dial code action: This option determined when VoIP dial code commands (DTMF / digit press) are performed:

Local caller: Dial code commands are performed only when the local side (see the setting Local IP Addresses).

Only that match the filter: Dial code commands are performed only they come from a phone that is in the recording filter.

All: Dial code commands are performed always.

Tenant: The dial code is performed only if coming from a phone of the tenant to which this call would be assigned.

Call pickup dial code: When a pickup dial code is specified (for example **), Apresa will attempt to detect such calls, and correct the remote number and direction, such that instead of, for example, **100, you see the real remote number. Apresa can only detect call pickup when it is started by dialing the call pickup dial code, followed by the extension. So if the dial code is **, and the extension is 100, then **100 is dialed. This feature is only available for the SIP protocol.

Alternatively, if the pickup code matches the complete phone number, for example, the destination phone number is "**333" and the pickup code is also "**333", then the direction is set to incoming, and no other changes are made. The actual phone number might be detectable using the "Use P-Asserted-Identity as identifier (Receiver)" setting.

Handle call legs separately: When a SIP call is forwarded with the same CallId through multiple hosts, each connection is handled as a separate call if this option is enabled. Otherwise, it is handled as a single call.

Use SIP tags for detecting end of call: This option is used for complex scenarios with multiple call legs. The SIP tags of some SIP fields are used to detect if all call legs are ended, or if the call continues. It is not recommended to switch this option on, unless necessary.

Detect Aastra Numbers: Enables detection of dialed telephone numbers when using an Aastra PBX.

H.323: Enable this option for Siemens HFA, Aastra / Ericsson Mx-One, and Avaya IP Office.

Avaya IP Office: This option, together with the H.323 option, must be enabled for Avaya IP Office H.323.

Local ID Detection (Avaya): Sets the method that will be used to detect the Local ID for the H.323-based Avaya protocol.

Avaya Stimulus: Enable this option for the SIP-based Avaya protocol used by IP Office.

Store on demand trigger text: If the specified text is detected on the phone display, then the call is stored, if store on demand is enabled. This option is for use with the Avaya SIP-based Stimulus protocol.

Detect queue number: If enabled, it will attempt to detect the queue number and place it in the Connected telephone number field (without the @host part). This option is intended for use with Avaya IP Office SIP.

Cisco Active Recording direction detection: When using Cisco active recording (built in bridge), the direction of the call (incoming or outgoing) can be detected based on the zero prefix of the remote telephone number. This zero prefix is removed from the telephone number, before display in the call listing. This might not work for internal calls, or when no dial prefix is used. Alternatively, the direction of the call can be detected using JTAPI. Fill in the server address of the Cisco PBX, and the username and password of the Application User created in the CUCM. For use of JTAPI, Java needs to be installed on Apresa.

Innovaphone: When using innovaphone PBX, and conferencing in calls to the Apresa VoIP Service, enable this option to detect the remote ID and direction of the main call.

Megaco / H.248: This option enables support for the Megaco protocol (also known as H.248).

MGCP: This is option enables support for the MGCP protocol, which is used by Panasonic and ShoreTel.

Nortel UNIStim: This option enables support for the Nortel Unified Networks IP Stimulus protocol.

Local ID: This option determines if the Local ID will be the detected telephone number, or the name (if available). (for Unistim)

Stop trigger texts (Alcatel): When one of the texts is found on the phone display, the recording is stopped and deleted. Multiple trigger texts can be specified separated by a comma.

OpenScape Xpert: This option enables support for recording the Unify OpenScape Xpert trading system. Recording also needs to be enabled and configured in the OpenScape Xpert System Manager.

Xpert Master Turret: The recorder will connect to this device to receive call meta data. To connect to multiple devices, use a comma separated list of IP addresses.

Xpert Local ID / Connected: These two settings determine which data items will be used to fill in the Local ID and Connected column in the database. This setting only effects new calls. Options:

- Line Name (Example: 5320). This is usually a line number. A turret can dial out and receive calls on multiple lines.

- Line Name (Example: 2.6.1/3.8). The identifier (ID) of the line.

- Username. This is the configuration name that is selected during log on.

- Device. The identifier of the device.

- Conference ID. This is the ID of the line conference, that is the same for all lines participating in the same conference on the same turret. It can have a different value on another turret.

SIPREC OpenScape Xpert: This options enables support for recording via the SIPREC protocol received from the OpenScape Xpert system. For this setting to work, it is also required that the VoIP Service is activated and configured for SIPREC. The OpenScape Xpert system also provides additional Xpert specific data about the recorded calls. This data can also be stored in the database. The following options control which information is stored where.

SIPREC Xpert Caller ID: This setting controls which data item is used to fill in the local, remote and connected caller ID columns in the database if the selected data option is available. The connected ID is only used when a conference is recorded.

Unique to this setting:

- Original. This uses the original number. In case a different selected data item is not available, this is also used.

SIPREC Xpert Data/SIPREC Xpert Notes: This setting controls which data item is stored in one of the notes (editable) or data (non-editable) columns in the database.

Unique to this setting:

- None. This leaves the column empty.

The following options are shared between both settings:

- Line Name. Line label of the affected Xpert Client.

- Line ID. The identifier (ID) of the line.

- Node Address. Xpert specific client ID.

- Agent Name. Profile name of the affected Xpert client.

- Device Type.

- Display Name. Mapping of the relevant Xpert Client (resolved by local telephone book or the display name coming from the SIP PBX)

The following data fields are available for SPM recording. There are up to four lines on each SPM panel that are assigned to a channel from 1 to 4.

These settings control how the data pertaining to the channels is stored in the database.

- SPM Channels. This stores all data from all channels in this field. The data is prefixed with the channel number and type.

- SPM Channel 1-4. This stores all data from this specific channel in this field. The data is prefixed with the type.

- SPM Line IDs. This stores the Line IDs from all channels in this field. The data is prefixed with the channel number.

- SPM Line ID 1-4. This stores the Line ID from this specific channel in this field.

- SPM Line Names. This stores the Line Names from all channels in this field. The data is prefix with the channel number

- SPM Line Name 1-4. This stores the Line Name from this specific channel in this field.

- SPM Caller Numbers. This stores the Caller Number from all channels in this field. The data is prefixed with the channel number.

- SPM Caller Number 1-4. This stores the Caller Number from this specific channel in this field.

- SPM Called Numbers. This stores the Called Numbers from all channels in this field. The data is prefixed with the channel number.

- SPM Called Number 1-4. This stores the Called Number from this channel in this field.

SIPREC Broadworks: Turns on support for Broadworks SIPREC extension data. Use the SIPREC Broadworks Data and SIPREC Broadworks Notes options to store the extra Broadworks data in the database notes or data fields. The following extra extension data is available:

- External tracking ID: Identifier for all recordings. Can be shared by multiple recordings if they were split due to call transfers.

- Service Provider ID: The identifier for the name of the service provider or enterprise to which the user requesting recording belongs.

- Group ID: The identifier of the group to which the user requesting recording belongs

- User ID: The identifier of the users for which the call is being recorded

- Call ID: The SIP call ID for the call being recorded

- Calling Party number: Phone number of extension that identifies the calling party

- Called Party number: The called party number after translations.

- Dialed digits: The digits dialed by the subscriber. Only available for outgoing calls.

- Redirection information: Information if a recorded call is redirected/transferred. Includes the phone number of extension of the redirected from party, the phone number or extensions of the redirected to party and potentially a new external tracking ID.

SIPREC Broadworks tenant assignment: Use this to select some extension data from Broadworks SIPREC. If the value matches a value configured in the settings for a tenant, the recording will be assigned to that tenant.

SIPREC Broadworks ignore store on demand: Broadworks extends the SIPREC protocol with a store on demand function. By default this extension will be detected if SIPREC broadworks support is turned on. If no store trigger is received during the recording, the recording will be discarded afterwards. Enabling this option will disable this functionality so that the recording will always be stored.

SIPREC Oracle: Turns on support for Oracle SIPREC extension data. Use the below options to store the extra extension data in the database notes or data fields or use it to determine call direction.:

- Request-Uri: The original Request-URI from the recorded call

- Realm: The original realm from the recorded call

- Direction Header: The Oracle SIPREC extension data allows for the copying of arbitrary SIP headers to the received metadata. With this option one such headers may be selected. If the value of that header matches the "Incoming" field, the call will be marked incoming in the database. If the value matches the "Outgoing" field, the call will be marked outgoing in the database.

- Header 1 to 6: The Oracle SIPREC extension data allows for the copying of arbitrary SIP headers to the received metadata. Use these options to configure up to 6 extra headers to should be searched for and to specify a database field in which the value of each header should be stored.

Use this to select some extension data from Oracle SIPREC. If the value matches a value configured in the settings for a tenant, the recording will be assigned to that tenant. If the option "Header" is chosen, fill in the Header field below with the specific header name that should be used.

Mitel MiNet: This option enables support for the MiNet protocol used by MiVoice Office and Mitel IP phones.

Samwin: This option enables support for the protocol used by Samwin, a call center product.

Local ID: This option determines if the Local ID will be the detected telephone number, or the UserID (if available). (for Samwin)

SIP NEC: This option enables support for the SIP NEC protocol.

SIPREC NEC: This option enables telephone number and direction detection for the NEC variant of SIPREC if this is received.

SIPREC use all transport addressess: This option needs to be enabled for some AudioCodes implementations of SIPREC, otherwise the phone numbers are not detected. Only enable if needed, because it diminishes general SIPREC compatibility.

SIP log-on detection: (Broadsoft) This option enables log-on agent detection for the hotelling function of Broadsoft.

SIP log-on detection: (Telephone number) This option enables log-on detection based on a telephone number that is dialed by the agent to log on or off.

Log on: Specify the telephone number that is dialed to log on. The agent id is expected to be dialed after this code.

Log off: Specify the telephone number that is dialed to log off.

Speakerbus: Enables integration with Speakerbus iSeries, using CDR data to detect phone numbers.

Port number: Apresa will accept TCP connections on the configured port, and interpret the received CDR Speakerbus data.

Apresa network service: This option needs to be enabled to record Microsoft Lync calls. It also requires that the Apresa Lync Plugin (ALP) is installed at the Lync server.

Filter Known Video: Enabling this option will filter out the RTP that is a known video codec used by Skype for Business. This will stop pure video sessions to show up as an unplayable recording in the calls database.

ALP IP addresses: The IP addresses on which the Apresa Lync Plugin will be installed and that will connect to this Apresa. This is used only for health monitoring. If connection to one of the ALP addresses is lost, this is regarded as a system error.

Click the Restart button to remotely restart the connected ALP instances (this functions independent of the filled in IP Addresses).

RTP over TCP: Checking this option allows for the recording of RTP audio streams that are sent over TCP instead of the usual UDP. This is very uncommon, but can occur with Skype for Business.

NAT Mapping: In case of a static NAT mapping between a public IP address and a private IP address, it may be possible that the public IP address is signaled, while the Apresa receives the RTP stream on the private IP address. In such situations, calls may not be properly recorded, because signaled information cannot be properly associated with the RTP streams, since the IP addresses differ. Using this option, a mapping can be made known to the Apresa, so that it can be used during call recording. Every mapping should be put on a new line with the form: <public IP address>=<private IP address>

Only record RTP in calls: RTP is the protocol used to transport audio between two parties over the network. RTP is used in call protocols such as SIP and H.323. If this option is enabled, and if no call signaling has been detected that is related to a detected RTP stream, then the RTP stream will not be recorded. Otherwise, it will be recorded, and the  stream will be added to the call database with IP addresses as identification, without telephone numbers.

Include port numbers in IDs when recording RTP outside of call: When recording RTP without any call signalling, enabling this option will add the recordings to the database with IP addresses and RTP port numbers as identification instead of just IP addresses

Invert reported port numbers when recording RTP outside of calls: This option only has an effect when recording RTP outside of a call and the option "Include port numbers in IDs when recording RTP outside of call" is enabled. When this option is disabled, the source IP and port are combined and the destination IP and port are combined into the ported caller ID. When this option is enabled the source IP and destination port and the destination IP and source port are combined into the reported caller IDs instead.

Amplification: The amplification of the VoIP audio data during recording. Use a positive value to increase the volume, or a negative value to decrease the volume.

Split long recordings: If a recording continues longer than the specified duration (in minutes), the recording is closed, and a new recording is started.

Maximum silence gap to include: When there is a gap in the audio stream of a call (a discontinuity in the RTP time stamps), and this gap is larger than the specified number of seconds, then this gap is removed from the recording, and a beep is inserted instead if the option for that is on. If not filled in, the default value is 10 seconds.

Insert tone at time jumps: A beep sound is inserted into the recording at the point where a gap was removed from the audio stream (a discontinuity in the RTP time stamps).

Collect information about all calls: This enables live information about calls even before they are connected (while ringing), and also about calls that are not recorded. It will enable a new menu item in the Tools menu called Active Calls (All).

Only enable when needed, because otherwise it uses resources unnecessarily.

Store metadata of missed calls: If enabled, missed calls are included in the call listing. This applies only to new calls.

Store metadata of calls over channel limit: If enabled, calls that were not recorded because of the channel license limit, are included in the call listing. This applies only to new calls.