Tools > VoIP Service

The VoIP Service allows the Apresa to function as a SIP phone, SIP server, or SIP proxy. It can accept multiple SIP calls simultaneously, and forward calls based on rules. This functionality can be used for:

•MoReSo (mobile recording)

•Cisco Dual Stream Active Recording

•Playback of a notification into the call

VoIP Service: This is the main switch that enables or disables this feature. If this setting is switched off, then no VoIP service is performed

Restart service: This is only needed in special situations, when the basic settings are changed. It causes active calls, that flow through Apresa, to be dropped. (See also Apply)

General

Delayed SDP offer: This option is related to a detail of the SIP protocol. It must be enabled for Cisco Dual Stream Recording.

Reject duplicated calls: Incoming calls with the same caller and receiver as an existing active call, are rejected. To let this option take effect, restart the VoIP service.

Accept SIP registrations: Enables other SIP phones or entities to register at Apresa. Details are configured under SIP Registrator.

NAT Address: Sets the IP address the VoIP services advertises for RTP connections. If left empty the local IP address will be used. This option can be used if the Apresa is behind a NAT and the local IP address can not be used. It may also be required to add the local to public IP mapping in the system settings for recording to succeed.

Maximum call duration: Any call that is active longer than the duration specified with this option will be stopped automatically. The maximum duration must be specified in seconds.

Maximum number of simultaneous calls: Any new call that would make the number of active calls exceed this value will be rejected automatically.

SIPREC: Enables handling of incoming SIPREC call data. Enabling this option also adds an additional SIP registration option (see below). It might be needed to configure the external telephony system to send SIPREC data to Apresa.

Determine direction: This SIPREC option can be used in combination with specifying local numbers on the system settings page under the VoIP setting to determine call direction. For this to work, the SIPREC client needs to provide the participant data in a predictable order. If one of the participants is detected as local and the other remote, this setting controls which participant is seen as the caller and which as the callee to determine if the call is incoming or outgoing. If the recording client does not provide the participants in a predictable manner, setting this option to disabled will mark the calls with unknown direction.

Protocol

With these options, transport protocols for SIP can be enabled. The local port on which the VoIP service expects the protocol can also be adjusted.

The options are UDP, TCP and TLS. SIP over TLS is encrypted. As TLS uses TCP, the TLS and TCP local port must not be the same if both are enabled.

TLS provides the following extra options:

Certificate: For the VoIP service to accept encrypted calls, a certificate is necessary. Certificates can be created or uploaded via on the certificates page. The list of all certificates that may be used for setting up the encrypted TLS connection for SIP calls. One may be selected here.

Verify certificates: When this option is enabled and a certificate is received from a TLS peer, the certificate is checked for its validity. This requires that the certificate is trusted. For outgoing TLS connections, this also requires that there is a subject alternative name in the certificate that matches the domain name that was contacted. This option will mainly apply for outgoing calls, but if a client sends a client certificate, this will be checked as well.

Record: Enable yes if encrypted calls must be recorded. Compatibility mode uses the recording method from older versions. It is not recommended for new installations and must not be used in combination with SIPREC.

If you enable TLS, also enable the Secure RTP option, otherwise the audio stream is not encrypted.

Codecs

Select the audio codecs to be enabled or disabled for the VoIP service. Only the enabled codecs will be accepted or offered for use in a SIP call.

All the listed codecs can be recorded by Apresa, but not all listed codecs can be used by the VoIP service to send audio. This is not a problem if the VoIP service only needs to accept calls, and when it does not need to send any audio.

The VoIP Service needs to be able to send audio when inserting a notification message or when forwarding a call when it stays in the loop.

In this case it is important that the selected codec is supported by the VoIP Service for sending audio. G.711, G.722, iLBC, and Opus audio can be sent, G.729 audio cannot be sent (but it can be received).

RTP

Minimum port and Maximum port: These settings control the range of UDP ports that the VoIP service will try to allocate for RTP sessions.

Inactivity timeout: This setting controls if a call is dropped by the VoIP Serivce when no RTP is received for a configurable timeout. By default no action is done. Select Start of Call to only check for inactivity before any RTP is received. After the first RTP is received, calls will then never be stopped even if the RTP stops later. Select Always to drop calls when at any point the RTP stops for the configured timeout

Timeout: This settings controls the timeout duration for the inactivity timeout. Specify the timeout in seconds. The minimum value is 1 second and the maximum value is 60 seconds

Secure RTP: This option will enable the use of encrypted RTP. This option should be used in combination with the TLS protocol for SIP.

SAVP: When this option is set, the VoIP service will generate SIP offers with the RTP/SAVP profile set when SRTP is on for outgoing calls or when the delayed SDP option is enabled. This may be required for interoperability. This setting only works when SRTP is enabled as well.

Last calls: This table provides a list of the calls that were handled by the VoIP service, and which rules was applied to them. This is useful when setting up and verifying rules.

SIP Line 1 / SIP Line 2: The VoIP service can communicate with possibly two SIP destinations (for example a SIP trunk, a SIP PBX, or a SIP phone).

IP name or IP address: The IP name or IP address of the SIP server.

SIP REGISTER: If enabled, Apresa will try to register itself with the specified username and password. This might be needed when communicating with a SIP PBX or SIP trunk.

Username: This will usually correspond to the telephone number of Apresa.

Password: The password used during SIP registration.

Domain: Apresa will register itself with username@domain. When left empty, the IP name or IP address of the SIP server, which is specified above, will be used as domain.

Local IP: This setting only needs to be set if a non-standard local port is defined (not 5060), and if the main IP address of Apresa should not be used as originating IP address. In that case, fill in the local IP address that should be used.

Local port: Fill in if Apresa needs to register itself on a non-standard local port (not 5060). This could be needed to avoid conflicts with other use of port 5060.

Registration interval: The SIP registration interval in seconds. When left empty, Apresa can use the default SIP registration interval, or the one advocated by the SIP server.

Local IP address in messages (NAT): If Apresa is communicating to a server in another network, it can be needed to specify the local IP address that Apresa must specify in SIP messages. If left empty, this will be determined automatically.

Protocol: Controls over which protocol the registration is made.

SIP Registrar: This section is available if the "Accept SIP registrations" option is enabled.

Username: The telephone number of the other entity that registers at Apresa

Password: The password that the other entity must provide to register at Apresa as this telephone number

To route calls to a registered telephone, use the Forward call action, and select Registered phone.

SIPREC: This section is available if the SIPREC option is enabled. Enable SIP REGISTER if Apresa needs to register itself as a SIPREC recording server. This is not needed for all SIPREC implementations.

Actions Rules: When an incoming call arrives, the system will verify if the call satisfies the conditions of rule 1. If so, then action of rule 1 is performed. Otherwise, the system will verify if the call satisfies the conditions of rule 2. If so, then the action of rule 2 is performed. Otherwise, the system will verify if the call satisfies the conditions of rule 3. And so on. Because at least one action rule has to be performed, the last rule has the condition "Always".

Condition:

•Always: The selected action will be performed unconditionally.

•Check source telephone number: The condition is satisfied if the telephone number of the initiator of the call matches any of the specified telephone numbers. Multiple telephone numbers can be specified separately, or using the wild cards * and ?, meaning * = any number of digits, ? = one digit.

•Check destination telephone number: The condition is satisfied if the telephone number of the receiver of the call matches any of the specified telephone numbers. Again the same wildcards are allowed.

Action: There are the following possible actions:

•Reject call: This causes the call to be terminated.

•Accept call: The call is answered. The system will keep the call active, until the remote side ends the call.

oPlay test tone: Enable this option to let the system playback test tone into the call. Otherwise, the system will remain silent. If the option "None" is chosen, it will not even send RTP packets; this can cause a time-out error on some systems.

•Forward call: The call is forwarded to one of the two SIP lines (defined earlier). When forwarding, there are the following options:

oForward destination: The SIP line to which to forward the call, or alternatively forward to registered phones (see SIP Registrar)

oProtocol: Controls over which transport protocol the SIP is forwarded. Original will use the same protocols for the incoming and outgoing call.

oDestination phone number:

▪Original: the destination telephone number is not changed, except that an optional dialing prefix can be removed from and/or added to the start of the phone number. This causes the system to behave like a SIP proxy.

▪Read phone number: This is for use with the MoReSo sim-chip.

▪Fixed: the call is forwarded to the custom fixed telephone number that is filled in

oSource phone number conversion: When forwarding a call to another SIP line, it might be needed or desirable to change what is reported as the telephone number of the originator of the call. This might be needed because the SIP line is a SIP trunk that allows only some source telephone numbers. Or secondly, it might be desirable, because it allows the final receiver of the call to see the right call-back number. The conversion table has the following format: Original number=New number. On the left hand side, in the original number, it is allowed to use * to match any phone number.

oStay in loop: When Apresa stays in the loop, this means all audio of the call must continue to flow through it, and bringing down Apresa, will cause the active calls to be dropped. When this option, "stay in the loop", is disabled, an attempt is made to stay out of the loop, but this is not guaranteed.

oUse username for caller: When forwarding to a SIP line, it will use the username of the SIP line registration as the source telephone number.

•Play notification message: An audio message will be played back into the call, and then the system will proceed with the next action. Click the Upload button to upload an audio file that contains the audio message that must be used as notification message. The audio file will be converted to an internal format for playback. If the audio file cannot be converted, first convert the audio file to a supported format (for example PCM .wav format).

oSend notification to: This option decides if the notification message will be heard by the caller or the receiver of the call.

•Selection menu: This allows the caller to make a choice using dial codes 0-9 or * or #. Upload an audio message to be played. Select the rule to jump to when a dial code is pressed by the caller. If no actionable dial code is pressed for the specified timeout period, then the timeout action will be performed. Without a timeout, the software will wait until the caller makes a choice or hangs up. The timeout can also be used to trigger repeat of the audio message, by selecting itself as the timeout action.

•Store this call: This signals to the recording component that the recording of the call must be stored, when store on demand is enabled.

•Start recording: This signals to the recording component that recording of the call must start, when recording on demand is enabled.

•Delete: This signals to the recording component that the recording of the call must be deleted. This also works when the Delete on demand option is switched off in the Recording settings.

Apply: New action rules and conditions can be applied without restarting, by clicking the Apply button. Active calls will not be dropped.