SIP trunk recording

A Session Initiation Protocol (SIP) trunk is a logical connection between an IP PBX and a service provider’s application server that allows voice over IP (VoIP) traffic to be exchanged between the two.

To deploy SIP trunks you need the following components:

  • PBX with a SIP-enabled trunk side

  • a SIP-compatible enterprise edge device (this can either be a firewall with complete support for SIP, or an edge device connected to the firewall handling the traversal of the SIP traffic)

  • and Internet Telephony Service Provider (ITSP) or SIP trunking service provider

When a call is placed from an internal phone to an external number, the PBX sends the necessary information to the SIP trunk provider, who establishes the call to the dialed number and acts as an intermediary for the call. All signaling and voice traffic between the PBX and the provider is exchanged using SIP and RTP protocol packets over the IP network.

If the called number is a traditional PSTN telephone, the trunk provider routes the IP packets to the PSTN gateway that is closest to the number being called, to minimize possible long distance charges. The provider can also terminate PSTN numbers, and route incoming calls for those numbers back to the IP PBX over the SIP Trunk. This allows businesses to offer local phone numbers in several geographical areas, but service them all from a single location.

If the called number can be reached over a SIP Trunk, the call does not need to be routed over the PSTN, but can instead be carried on the IP network end-to-end, creating a very cost-effective solution. SIP trunking can also serve as the starting point for the entire breadth of real time communications possible with the protocol, including Instant Messaging, presence applications, white boarding and application sharing.

The SIP trunk can be provided by the Internet Service Provider (ISP), or by an independent ITSP. In fact, there can be several parties involved, each one providing a different part of the service required to deliver end-to-end communication.

Because a SIP trunk is not a physical connection, there is no explicit limit on the number of calls that can be carried over a single trunk. Each call consumes a certain amount of network bandwidth, so the number of calls is limited by the amount of bandwidth that can flow between the IP PBX and the provider’s equipment.

The Recorder records traffic at the SIP Trunk. This includes environments in which SIP trunk sessions are replicated by an edge device such as Acme PacketTM SBC to the Recorder.

The way in which traffic is provided to the Recorder depends on the spanning/replication mode. In SIP Trunk Recording, the edge device provides the Recorder with both signaling and audio; in this case, the signaling does not carry the employee’s extension. SIP Trunk Recording is therefore established at the member group level (not at the extension level). A SIPREC adapter created in Recorder Manager allows the retrieval of custom tags Time-stamped information items appended to interactions at different points of interest, including Annotations and Events. from SIP headers or SIPREC metadata.

You may have multiple data sources in environments in which there are multiple tenants (contact The entire communication experience for a customer, from beginning to end. centers) hosted on separate switches, and each tenant is identified by a unique switch IP address.

Each switch requires its own Phone data source Third-party systems that provide data to the system, including employee and device states, and data change events. Typical data sources are phone switches, PBXs, or LANs., with the switch IP address or host name configured as the call center's SIP trunk interface. The Server Type under Settings > Device Configuration should be set to PSTN Side - Far End. The Recorder uses this IP address or host name to identify the data source and tag it to the recordings. The Recorder Integration Service uses the data source tagged to the recordings to then match it to relevant CTI.

For configuration details see Gateway side correlation pool member group settings, and the Avaya or Genesys Integration Guide. (You may have multiple data sources that use this member group type, but only one member group per data source.)

Calls are not tracked across data sources, and therefore are not tracked across different switches.

This limitation means that if a call is transferred between switches it will not be tracked from one switch to the next, and will therefore be considered two separate calls.

SIP Trunk Recording works with all of the following recording features and modes, and no special configuration:

  • Application, Performance and Liability Recorder Fallback Types

  • Shared Interception and Dedicated Interception Load Balancing Types

  • VOX Fallback

  • Redundancy

  • Screen Recording

  • Recording in SIP and TDM mixed trunk environments

  • Recording SIP trunk traffic from multiple PBX/ACDs

  • Option to stop recording when the Customer is put on hold

Related topics 

Recorder reference