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Configuration

This topic contains detailed configuration guidelines to enable call recording for SIPREC:

Step

Reference

Description

1

SRC Configuration

Configure the SRC (Session Recording Client) for call recording.

Note

This configuration should be performed by the customer’s Telephony Engineer.

2

Integration Adapter Configuration

Configure the Integration Adapter on the Collector.

3

RTP Collector Configuration

Configure the RTP Collector on the Collector.

4

SIP RAM Configuration

Configure the SIP RAM on the Collector.

5

Configure Custom Metadata Extensions

Configure Custom Metadata Fields on the Collector.

SRC Configuration

SRC (Session Recording Client) configuration will vary depending on the telephony platform being used. Please refer to the telephony platform documentation.

Integration Adapter Configuration

The Integration Adapter configuration is specified in IntegrationAdapter.ini and Identity.ini. These files are automatically generated with default values when the Collector is started for the first time.

IntegrationAdapter.ini
Example
[General]
DownstreamAPIVersion=0
RecordByDefault=1

[JetStream]
ServerURL=nats://u-capture.example.com:4222

[VoiceActivityDetection]
WindowSizeMS=10
UseShortTermEnergy=1
MinEnergy=0.000060
UseZeroCrossingRate=0
MinZeroCrossingRate=0.000001
MaxZeroCrossingRate=0.187400
UseActivityTriggering=0
MinActivityMS=20
ActivityEndHangTimeMS=5000

[Licence]
RecordPoolChannels=1000
PP0=P9XCS8KLATSJNHGN

[INSTALLED PPS]
PP0=/usr/lib/libPP_Matcher.so

[Phones]
DetectNewItems=1
Settings

Only configuration settings relevant to this SIPREC integration are listed here. For all other settings, leave these set to their default values.

Note

2024.2 introduced the support of NATS JetStream, therefore the NATS Streaming settings (STAN and STANChannels) have been depreciated. Uniphore recommends using the new JetStream settings however the depreciated NATS Streaming settings are still available. For the depreciated STAN settings and an example of those settings in context see Depreciated NATS Streaming Settings.

[General]

[General]DownstreamAPIVersion

Controls the format of messages sent downstream to consuming applications, either:

  • Set to 0 (default) to use the original message format of metadata for metadata, and media.<call_id> for media.

  • Set to 1 to use the format call.metadata.tenant.<tenant_id>.call.<call_id> for metadata and call.media.tenant.<tenant_id>.call.<call_id> for media.

[General]RecordByDefault

Set to 1 to enable or 0 to disable recording by default, for newly discovered devices. If not specified, the default value is 1 (enabled).

[JetStream]

[JetStream]ServerURL

The URL of the NATS server.

[JetStream]Username

If the NATS server is configured for user based authentication, specify the username here. If not specified, no value is used.

[JetStream]Password

If the NATS server is configured for user based authentication, specify the password here. If a password is used, the NATS server should also be configured to use TLS to prevent the password from being sent over the network in the clear. If not specified, no value is used.

[JetStream]Token

If the NATS server is configured for token based authentication, specify the token here. If a token is used, the NATS server should also be configured to use TLS to prevent the token from being sent over the network in the clear. If not specified, no value is used.

[JetStream]TrustedCACertificatesFileName

If the NATS server is configured to use TLS, specify the trusted CA certificate chain here. This will be the CA certificate chain that was used to sign the server certificate. If not specified, no value is used.

[JetStream]ClientCertificateChainFilename

If the NATS server is configured to use TLS with mutual-authentication, specify the client certificate chain here. This will be the client certificate plus the CA certificate chain used to sign the client certificate. If not specified, no value is used.

[JetStream]ClientCertificateKeyFilename

If the NATS server is configured to use TLS with mutual-authentication, specify the client private key here. If not specified, no value is used.

[JetStream]SkipServerVerification

If the NATS server is configured to use TLS, disables verification of the server certificate. Not recommended. If not specified, the default value is 0.

[VoiceActivityDetection]

[VoiceActivityDetection]

This section of the ini file is used to define the VAD settings. If the default configuration values don’t work for the customer, it’s recommended that you disable VAD, then record and export some example calls to WAV. These files can then be passed through a test tool that will allow the trial of different configuration values to ensure the correct parts of the call are identified as voice. The best fit configuration can then be applied to the Collector. Please contact the Uniphore Development Support Team for details.

To disable Voice Activity Detection, simply disable the use of both Short-Term Energy and Zero Crossing Rate.

[VoiceActivityDetection]WindowSizeMS

The size of the window, in milliseconds, on which to perform VAD. The default is 10 milliseconds – we don’t recommend any other value at present.

[VoiceActivityDetection]UseShortTermEnergy

Enables/disables use of the Short-Term Energy algorithm. By default, this is enabled (1).

[VoiceActivityDetection]MinEnergy

The minimum energy required for the window to be considered as voice activity. Expressed as a decimal value between zero and one. Zero being absolute silence and one being maximum loudness. The default is 0.000060. This may seem like a low value, but generally background noise will be lower than this.

[VoiceActivityDetection]UseZeroCrossingRate

Enables/disables use of the Zero Crossing Rate (ZCR) algorithm. Use of ZCR is only recommended if it’s required that tones on channel do not trigger voice recording. If this is the case, ZCR should be used in conjunction with STE. By default, ZCR is disabled (0).

[VoiceActivityDetection]MinZeroCrossingRate

The minimum Zero Crossing Rate for the window to be considered as voice activity. The default is 0.000001. Human speech has a low ZCR, increasing this value may prevent voice from being detected. A lower limit is required because a ZCR of zero indicates that the sine wave is not crossing zero i.e. the waveform is flatline (silence).

[VoiceActivityDetection]MaxZeroCrossingRate

The maximum Zero Crossing Rate for the window to be considered as voice activity. The default is 0.187400.

[VoiceActivityDetection]UseActivityTriggering

Enables/disables use of activity triggering. By default this is disabled (0). If enabled (1), a voice recording will not be created until voice activity is detected on the channel. Likewise, the voice recording will be ended on cessation of voice activity on the channel. If disabled (0), voice recordings will be created and ended based on signalling as normal.

[VoiceActivityDetection]MinActivityMS

The minimum duration of voice activity, in milliseconds, that is required to invoke voice recording. The default is 20 milliseconds. This value must be a multiple of the window size and must be between 10 and 300 milliseconds.

[VoiceActivityDetection]ActivityEndHangTimeMS

The duration, in milliseconds, to continue the voice recording after voice activity has ceased. The default is 5000 milliseconds (5 seconds). This value must be a multiple of the window size. This prevents pauses in speech from causing excessive stopping and restarting of recording.

[Licence]

[Licence]RecordInternalCalls

Enable (1) or disable (0) the recording of internal calls (extension to extension calls). If not specified, the default value is 1.

[Licence]RecordPoolChannels

Defines how many concurrent channels can be captured by the Collector. Calls over this value will be discarded. If not specified, the default value is 1000.

[Licence]PP<0…N>

PP license code for the Collector. This will have been provided as part of the customer’s deployment information from Uniphore.

[Phones]

[Phones]SegLengthSecs

The maximum recording segment length in seconds. After this duration, recordings will be split. If not specified, the default value is 3600 (1 hour).

Note

For call segmentation to work in Communication Recording Agent, SegLengthSecs should be set to an extremely high value (e.g. 9999999). This will prevent call segmentation from being carried out by the Collector.

[Phones]DetectNewItems

Enable (1) or disable (0) the automatic detection (discovery) of new devices. The default is enabled (1). In general, leave this set to 1 (enabled) for initial system setup and device discovery – if the customer then wants to “lock down” their recordable devices, this can be changed to 0 (disabled).

[Recording]

[Recording]MaxPacketsInQueue

The maximum number of packets to queue for processing before discarding packets. If not specified, the default value is 65536.

[Recording]MaxPacketLength

The maximum packet length to process in bytes. If not specified, the default value is 9018 (18 bytes for the Ethernet header and FCS, plus a 9000 byte MTU).

[Recording]MediaFlushTimeoutMS

The maximum interval (in milliseconds) to flush media data to NATS Streaming. If not specified, the default value is 10000 (10 seconds). Realtime streaming solutions should set the value to 100.

[Recording]Triggers

Set the comma separated list for the priority of recording triggers, valid options are CTI, Activity, and Signalling.

Under normal operation, the first trigger listed is used – if the source of that trigger fails then the next working trigger source in the list is used to determine when to trigger recording.

[Recording]DelayedCallEndTimeoutMS

Determine how long (in milliseconds) to continue capturing each call after the call end signal has been received, this is to allow any remaining audio to be captured. If not specified, the default value is 1000 (1 second)

[Recording]Policy

Determine which calls should be recorded, either internal extension to extension calls (Internal), external calls (External), or both internal and external calls (All). If not specified, the default value is to record just external calls (External).

Note

This setting replaces the now depreciated [Licence]RecordInternalCalls setting.

[CallSuppression]

[CallSuppression]MaxSuppressCount

The maximum number of times a given recording can be suppressed. If not specified, the default is 3 times (3).

[CallSuppression]MaxSuppressTime

The maximum length of time a given period of suppression will last. If not specified, the default is 15 seconds (15).

[CallSuppression]StopCapture

When call capture is suppressed, send (true) or do not send (false) a request that the audio feed from the 3rd party system be stopped. If not specified, the default is to send the request (true).

[CallSuppression]StopOnCallEnd

Determine if the suppression for a device is ended when the suppressed call on that device ends. If not specified, the default is false.

Depreciated NATS Streaming Settings

[STAN]ServerURL

Set to the URL of the NATS Streaming Server associated with this Collector.

[STAN]ClusterID

Defines the NATS Streaming cluster ID – must be set to stan-cluster.

[STAN]ClientID

A string that identifies this NATS Streaming client. The Collector ID is appended to this to ensure it is unique. If not specified, the default value is RedBox-IntegrationAdapter.

[STAN]Username

If the NATS Streaming Server is configured for user based authentication, specify the username here. If not specified, no value is used.

[STAN]Password

If the NATS Streaming Server is configured for user based authentication, specify the password here. If a password is used, the NATS Streaming Server should also be configured to use TLS to prevent the password from being sent over the network in the clear. If not specified, no value is used.

[STAN]CertificateChainFilename

If the NATS Streaming Server is configured to use TLS, specify the trusted CA certificate chain here. This will be the CA certificate chain that was used to sign the server certificate. If not specified, no value is used.

[STAN]ClientCertificateChainFilename

If the NATS Streaming Server is configured to use TLS, specify the client certificate chain here. This will be the client certificate plus the CA certificate chain used to sign the client certificate. If not specified, no value is used.

[STAN]ClientCertificateKeyFilename

If the NATS Streaming Server is configured to use TLS, specify the client private key here. If not specified, no value is used.

[STAN]Ciphers

If the NATS Streaming Server is configured to use TLS, this value can be used to restrict the allowed ciphers. If not specified, no value is used.

An example of the depreciated NATS Streaming settings:

[STAN]
ServerURL=nats://abcd.dev.mycompany.com:4222
ClusterID=stan-cluster
ClientID=RedBox-IntegrationAdapter-1
ConnectionTimeoutMS=15000
RetryWaitMS=15000
Username=
Password=
SkipServerVerification=0
CertificateChainFilename=
ClientCertificateChainFilename=
ClientCertificateKeyFilename=
Ciphers=

[STANChannels]
Alerts=RedBox.Collector.Alert
EventLogs=RedBox.Collector.EventLogs
ConfigRequests=RedBox.Collector.ConfigRequests
ConfigEvents=RedBox.Collector.ConfigEvents
Media=media
Metadata=metadata
Identity.ini
Example
TenantID=dcedc429-11a9-4408-84a7-c781363c62e1
TenantName=test
CollectorID=0f89817f-e324-48ca-bcb6-3d61f292755b
CollectorName=collector1
Settings

TenantId

Set to the unique identifier of the tenant associated with this Collector. If no tenants are defined, the default test tenant will be used dcedc429-11a9-4408-84a7-c781363c62e1

TenantName

The name of the tenant to which the Collector belongs (optional).

CollectorId

The unique identifier for this Collector. A UUID will be automatically generated and written to Identity.ini on first start.

CollectorName

The name of the Collector (optional).

RTP Collector Configuration

The RTP Collector works in sync with the SIP RAM. The SIP RAM manages the SIP dialog and the RTP Collector owns the RTP session.

The RTP Collector configuration is specified in RTPCollector.ini – this file is automatically generated with default values on first start. An example is shown below:

[MediaPorts]
Min=25000
Max=65535

[SRTCP]
UseSRTCPWhenSRTP=1

[docker0]
Use=0
IPv4Address=10.0.2.15

[<adapter-name>]
Use=1
IPv4Address=10.0.2.42
IPv6Address=2001:db8::1234:5678
Settings

Only config settings relevant to this SIPREC integration are listed here. For all other settings, leave these set to their default values.

[MediaPorts]

Defines the RTP media port range to capture data from.

[MediaPorts]Min

The beginning of the media port range. Default is 25000.

[MediaPorts]Max

The end of the media port range. Default is 65535.

[SRTCP]UseSRTCPWhenSRTP

Enables/disables the expectation that RTCP will also be encrypted when SRTP is being used for a given session. If not specified, defaults to 1 (enabled).

[<adapter-name>]

The RTP Collector discovers all available network adapters on start-up – they’re listed by their device name (GUID). The adapter’s IPv4 address is displayed for ease of identification. By default, all adapters are enabled for media traffic. Enable (Use=1, default) or disable (Use=0) adapters as needed.

It’s important that only adapters reachable by the telephony system are used. Adapters are load balanced, so sessions will be evenly distributed across all enabled adapters.

[<adapter name>]AdvertisedIPv4Address

Overrides the IPv4 address advertised for the given network adapter. This is useful when you need an ingress address that is not an address belonging to the RTP Collector itself. If not specified, then defaults to the adapter’s IPv4 address.

SIP RAM Configuration

The SIP RAM configuration is specified in SIP_RAM.ini – this file is automatically generated with default values on first start. An example is shown below:

[SIPRAMSettings]
Protocol=udp
SIPPort=5060
MaxConcurrentCalls=1000
ExtensionField=from

[SIP]
MessageLogging=1

[SecureConnections]
Ciphers=
ClientVerificationMode=Mandatory

[SRTP]
Disabled=0
Mode=2

[RTPCollector1]
Host=localhost
Port=6500
TimeoutMS=1000

[Control]
ListenPort=6501

[Codecs]
Count=23
0=G.711 mu-law
1=GSM 06.10723.1
3=ADPCM 32 kbit/s
4=ADPCM 64 kbit/s
5=G.711 a-law
6=G.722
7=G.728
8=ADPCM 44.1 kbit/s
9=ADPCM 88.2 kbit/s
10=G.729
11=G.722.1 24 kbit/s 16 Khz
12=G.722.1 32 kbit/s 16 Khz
13=G.722.1 24 kbit/s 32 Khz
14=G.722.1 32 kbit/s 32 Khz
15=G.722.1 48 kbit/s 32 Khz
16=G.726 16 kbit/s
17=G.726 24 kbit/s
18=G.726 32 kbit/s
19=G.726 40 kbit/s
20=AMR Narrow-Band
21=AMR Wide-Band
22=Opus

[SIPRegistration1]
AoR=
Password=
OutboundProxy=

[SIPREC]
ExtensionModuleDirectory=/usr/local/lib/SIPREC
Settings

Only config settings relevant to this SIPREC integration are listed here. For all other settings, leave these set to their default values.

[SIPRAMSettings]IP

The hostname or IP address of the Integration Adapter. If not specified, defaults to localhost.

[SIPRAMSettings]Protocol

The SIP transport protocol to use (udp, tcp, tls, or any). Default is udp.

[SIPRAMSettings]SIPPort

The port to listen for SIP communications on. Default is 5060. If [SIPRAMSettings]Protocol is an explicit protocol, then [SIPRAMSettings]SIPPort describes an explicit port for that protocol. If [SIPRAMSettings]Protocol is any, then [SIPRAMSettings]SIPPort describes the base port used for non-secure SIP and secure SIP uses the base port plus one.

[SIPRAMSettings]MaxConcurrentCalls

The number of active SIP sessions allowed before new sessions will be rejected with a SIP 503 response. Default is 1000. This setting should match the channel count - see [Licence]RecordPoolChannels.

[SIPRAMSettings]ExtensionField

Allows the SIP RAM to be configured to take the user from the to header as the recorded extension. If not specified, defaults to from.

[SIP]MessageLogging

Enables (1) or disables (0) logging of SIP messages. If not specified, defaults to 1.

[SecureConnections]Ciphers

Allows the TLS and DTLS ciphers to be overridden.

[SecureConnections]ServerCertificatePath

The path to the server certificate that identifies the Uniphore SIP User Agent. This is only required where TLS or DTLS transports are in use. The certificate file must be PEM encoded.

[SecureConnections]ServerKeyPath

Specifies the PEM format private key file that corresponds to the SIP RAM server certificate. If not specified, no certificate is used.

[SecureConnections]ClientVerificationMode

Determine the client certificate verification behaviour. Valid values include Optional (Default, the client certificate is validated if the client presents one. If no certificate is sent, the connection is still accepted), None (The client certificate verification is completely disabled. Invalid or missing certificates are ignored), and Mandatory (The client must present a valid certificate. The TLS handshake is rejected if no valid certificate is provided).

This setting must be set to None if the client sends a certificate.

[SRTP]Disabled

Disables use of SRTP (set to 1). If not specified, defaults to 0.

[SRTP]Mode

Specifies the negotiation mode for setting up SRTP. If not specified, defaults to 2 (best effort). Acceptable values are:

1 (mandatory – if the peer does not support SRTP, the session will fail)

2 (best effort – if the peer supports SRTP, SRTP will be used)

3 (fallback)

4 (multiple media descriptor)

[ RTPCollector<1…N>]Host

The hostname or IP address of the RTP Collector. If not specified, defaults to localhost.

[RTPCollector<1…N>]Port

The port on which to communicate with the given RTP Collector. If not specified, defaults to 6500.

[RTPCollector<1…N>]TimeoutMS

The duration, in milliseconds, to wait for a response from the given RTP Collector before considering the request timed out. If not specified, defaults to 1000.

[ Control]ListenPort

The port on which to listen for client connections from services wishing to control the SIP RAM. If not specified, defaults to 6501.

[Codecs]

This section contains the enabled codecs in priority order. Values must be entered as shown below (case and whitespace sensitive) – possible values are as follows:

  • G.711 a-law

  • G.711 mu-law

  • G.722

  • G.722.1 24 kbit/s 16 Khz

  • G.722.1 32 kbit/s 16 Khz

  • G.722.1 24 kbit/s 32 Khz

  • G.722.1 32 kbit/s 32 Khz

  • G.722.1 48 kbit/s 32 Khz

  • G.723.1

  • G.726 16 kbit/s

  • G.726 24 kbit/s

  • G.726 32 kbit/s

  • G.726 40 kbit/s

  • G.728

  • G.729

  • ADPCM 32 kbit/s

  • ADPCM 44.1 kbit/s

  • ADPCM 64 kbit/s

  • ADPCM 88.2 kbit/s

  • AMR Narrow-Band

  • AMR Wide-Band

  • GSM 06.10

  • Opus

[SIPRegistration<N>]AoR

The Address of Record to register. Should be of the form:

<schema>:<user>@<domain>[:<port>][;transport=<protocol>]

Where:

  • Schema is either sip or sips.

  • User is the user to register.

  • Domain is the domain to which the user belongs.

  • Port (optional) default: 5060 for udp and tcp, 5061 for tls and dtls. An integer between 1 and 65535.

  • Protocol (optional) default: udp if [SIPRAMSettings]Protocol=any , otherwise the specified protocol. Acceptable values are udp, tcp , tls, and dtls.

[SIPRegistration<N>]Password

The password to use when presented with an authentication challenge. Will be removed on first start and the value encrypted and stored in [SIPRegistration<N>]Password.Encrypted.

[SIPRegistration<N>]Password.Encrypted

The encrypted password. Do not modify.

[SIPRegistration<N>]OutboundProxy

The hostname or IP address of the outbound proxy (optional).

Configure Custom Metadata Extensions

Since SIPREC metadata XML varies between telephony platforms, we recommend that you take a Wireshark trace or review SIP messages.txt and examine the XML data to analyze the structure and available fields. A sample set of recording metadata is shown below:

<?xml version="1.0" encoding="UTF-8"?>
<recording xmlns='urn:ietf:params:xml:ns:recording:1'>
  <datamode>complete</datamode>
  <group group_id="NzI0MDc5MDAtYmI2ZS0xMA==">
    <associate-time>2018-10-26T16:59:38Z</associate-time>
    <callData xmlns='urn:ietf:params:xml:ns:callData'>
      <fromhdr>
        "01159377100" &lt;sip:01159377100@172.16.250.4:5060&gt;;tag=gK082281ca
      </fromhdr>
      <tohdr>
        &lt;sip:01483920633@192.168.1.13:5060&gt;;tag=0_3017003168-335468980
      </tohdr>
      <callid>524346_134154658@172.16.250.4</callid>
      <gcid>524346</gcid>
    </callData>
  </group>
  <session session_id="NzQ0MTQxM2UtYmI2ZS0xMA==">
    <group-ref>NzI0MDc5MDAtYmI2ZS0xMA==</group-ref>
    <start-time>2018-10-26T16:59:38Z</start-time>
  </session>
  <participant participant_id="NzI0MDc5MDEtYmI2ZS0xMA==">
    <nameID aor="01159377100@172.16.250.4:5060">
      <name xml:lang="en">01159377100</name>
    </nameID>
  </participant>
  <participant participant_id="NzI0MDc5MDItYmI2ZS0xMA==">
    <nameID aor="01483920633@192.168.1.13">
      <name xml:lang="en"> </name>
    </nameID>
  </participant>
  <stream stream_id="NzI0MDc5MDQtYmI2ZS0xMA=="
      session_id="NzQ0MTQxM2UtYmI2ZS0xMA==">
    <label>1</label>
  </stream>
  <stream stream_id="NzI0MDc5MDUtYmI2ZS0xMA=="
      session_id="NzQ0MTQxM2UtYmI2ZS0xMA==">
    <label>2</label>
  </stream>
  <sessionrecordingassoc session_id="NzQ0MTQxM2UtYmI2ZS0xMA==">
    <associate-time>2018-10-26T16:59:38Z</associate-time>
  </sessionrecordingassoc>
  <participantsessionassoc participant_id="NzI0MDc5MDEtYmI2ZS0xMA=="
      session_id="NzQ0MTQxM2UtYmI2ZS0xMA==">
    <associate-time>2018-10-26T16:59:38Z</associate-time>
  </participantsessionassoc>
  <participantsessionassoc participant_id="NzI0MDc5MDItYmI2ZS0xMA=="
      session_id="NzQ0MTQxM2UtYmI2ZS0xMA==">
    <associate-time>2018-10-26T16:59:38Z</associate-time>
  </participantsessionassoc>
  <participantstreamassoc participant_id="NzI0MDc5MDEtYmI2ZS0xMA==">
    <associate-time>2018-10-26T16:59:38Z</associate-time>
    <send>NzI0MDc5MDQtYmI2ZS0xMA==</send>
    <recv>NzI0MDc5MDUtYmI2ZS0xMA==</recv>
  </participantstreamassoc>
  <participantstreamassoc participant_id="NzI0MDc5MDItYmI2ZS0xMA==">
    <associate-time>2018-10-26T16:59:38Z</associate-time>
    <send>NzI0MDc5MDUtYmI2ZS0xMA==</send>
    <recv>NzI0MDc5MDQtYmI2ZS0xMA==</recv>
  </participantstreamassoc>
</recording>

Create a Custom.ini file – an example is shown below:

[CallID]
Location=group.callData.callid
Regex=^[\w-]+
FieldId=96

[GCID]
Location=group.callData.gcid
FieldId=1000

In this example, the value of the group.callData.callid element, up until the ‘@’, will be populated into field ID 96 (AnnotationText1) and the value of the group.callData.gcid element will be populated into field ID 1000 (AnnotationText2).

Each configured item takes the following format:

[<Name>]
Location=<Location>
Regex=<Regex>
FieldId=<DatabaseFieldID>

Item

Description

<Name>

Name of the configured value. Must be unique in the Custom.ini file.

<Location>

The XML path to the element of interest. Relative to recording. Case sensitive.

Where multiple instances of an element exist, an array index may be specified. If no index is specified, index zero is assumed. E.g. participant[1]. nameID[0]. name[0].

<Regex>

Optional. Required if it is desirable to capture only a part of the element value. A standard regular expression. Use of www.Regex101.com is advised.

<DatabaseFieldID>

The ID of the database field that should be used to store this value. Field ID’s are listed in Database Field IDs. The selected database field can be mapped to an appropriate Metadata field using the Communication Recording Agent mapping features.

The SIP RAM service or Collector must be restarted for changes to this configuration to take effect.

SIPREC Recording Control

  • The recordable item is configured for recording in the SRC and then a call has been made.

In this section: