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Pulse System Description

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Pulse System

Overview

The Vingtor-Stentofon Pulse system is a SIP intercom system intended for small to mid-size installations for up to 64 IP stations. Some of the key features of Pulse are:

  • Easy to install and configure
  • Cost effective for solutions where up to 64 IP stations are needed in the network
  • Supports creating Pulse trunks to inter-connect up to 50 different Pulse systems
  • Single Pulse system works across multiple network subnets
  • SIP Gateway interface that is easy to setup (e.g. SIP-to-GSM, SIP-to-PSTN, etc.)
  • Open IP standards (SIP, HTTP(S), NTP, SNMP, RTP, etc.)
  • Shared network infrastructure with other systems (CCTV, PCs, etc)
  • Special SDK with API that can be used to create custom intercom solutions
  • Integrated managed switch in terminals
  • Wide set of special purpose IP stations and PC client
  • VS-IMT: Dedicate management tool for easy configuration of large number of stations

Pulse server functions

Specifications

Call capacity
Max no Vingtor-Stentofon terminals 64
Max no 3rd party terminals 10
Max no VS-Clients 10
Max no two way calls 32
Max no group calls 4
Call setup time two way calls 100 ms
Call setup time all call 300 ms
Operation and maintenance
Autodiscovery Yes
Centralized management Yes
Encrypted communication Yes
Integrated web server HTTP(S)
Software download TFTP
Network clock NTP
Multiple language Yes
Admin menu Yes
User menu Yes
External communication
Interface Analogue/GSM
SIP Gateways
Audiocodes MP-114 Analogue FXO
Grandstream HT503 Analogue FXO
Portech MV-370 GSM
SIP telephones
Grandstream GXP1625, GXV3240, GXV3275
SNOM SNOM870, SNOM821
Softclients X-lite, Bria
Cisco Cisco 9971
Yealink VP530, T48G, T49G
VoIP protocols and functions
SIP support RFC 3261 (SIP base standard)
RFC 3215 (SIP REFER)
RFC 2976 (SIP INFO)
DTMF support RFC 2833, 2976 (SIP info)
SDP RFC 4566
RTP RFC 3550
Audio functions
Adaptive Echo Cancellation (AEC) Yes
Noise Reduction (NR) Yes
Automatic Volume Control (AVC) Yes
Microphone sensitivity Configurable
Volume control Yes
Push to talk mode Yes
Open duplex (AEC) Yes
Switched duplex (AES) Yes
Voice codec G.722 (HD Voice)
G.711 A and µ (Telephony)
G.729
Jitter buffer Adaptive
IP networking and security
2 port data switch
Supplementary functions
Call hold Yes
Call transfer Yes
Call forwarding Yes
Group call Yes – 4 groups and priorities
Ring list Yes – parallell calling
Busy override Yes
Remote control Yes
Group Call answer Yes
Group Call w/ Recall Yes
Call queuing Yes
Network Multi-Subnet support Yes


Protocols and network ports

Protocol name TCP / UDP Port Description
SSH tcp 22
TFTP Server udp 69
HTTP tcp 80
SNMP udp 161
HTTPS tcp 443
SIP tcp 5060
SIP udp 5060
SIPS tcp 5061
Pulse udp 5062
mDNS udp 5353
ZapWeb tcp 8080
DIP tcp 50001
DIP Multicast udp 50001
Discovery udp 50002
ZAP tcp 50004
Demo tcp 50010
VoIP udp 61000:61150 VoIP port range is configurable on StationWeb. Note: only works for SIP mode


Reserved IP addresses and ports

Service TCP / UDP Address / Port Description
Group Call control UDP 239.192.2.0 :50001
Group Call audio UDP 239.195.0.0/16 :61060-61066 Server will generate a set of multicast addresses to use based on it's MAC address
Station Auto-discovery UDP 239.192.0.199 :50002 Station auto-discovery uses both broadcast and multicast


Dialing method

Pulse uses overlap dialing method. Overlap signaling means that digits are sent one by one from the station to the server (and not collected in the station and sent as a block, known as En Bloc dialing). Using overlap signaling, call setup can begin before all the digits have been collected.

Gateway interface and external communication

The Pulse server can be configured with up to 10 Gateway accounts. Appropriate Pulse-Server-TelTrunk license allows user to create one GSM or telephone trunk channel in a Pulse Server; the Gateways parameter will be available under Server Management -> Server Configuration -> Directory Settings. For every gateway a separate SIP Gateway account needs to be configured.

Adding a SIP Gateway account


Directory Number 0 is optional as you can use any number to make external calls through the Gateway.

Signal diagram


Configuring the Telephone Gateway is done by logging into the Telephone Gateway to register it to the Pulse Server by using the created SIP Gateway Account.

Currently supported Gateway devices are and their configuration is described in separate articles:

Support for 64 stations

A single Pulse system supports up to 64 registered Vingtor-Stentofon IP stations. The auto-discovery feature makes it easy to configure larger Pulse installations using only station built-in web services.

Pulse will allow for 16 IP stations to register out-of-box without any licenses. The system can be further expanded by installing an IP Station license for each additional intercom. Pulse licensing is described in detail in SIP & Pulse License management.

Station auto-discovery and configuration

All Vingtor-Stentofon IP stations use auto-discovery feature which enables fully automatic discovery of all the IP stations on then network by Pulse server.

Station configuration can be done directly from Pulse server StationWeb. After IP stations are discovered with Pulse station auto-discovery, they can be immediately set up and configuration saved and applied for multiple stations at once. All the basic functions can be directly configured from Pulse server. For more specific configuration and customization, StationWeb of every IP station can be used as configuration tool. In addition, VS-IMT tool can be used to configure one or multiple stations in Pulse system.

More info can be found in Station auto-discovery and Configuration (Pulse).

Multi-subnet support

Pulse can be deployed as a single-server system with IP stations being installed in different network subnets. There are some prerequisites for network setup so Pulse can operate without problems:

  1. Network subnets (segments) with IP stations need to be routable
  2. IP Multicast routing needs to be enabled and configured on the layer-3 switch or router. This is required by Pulse IP station auto-discovery and Pulse Group Call functions
  3. All protocols and ports that Pulse IP stations are using must not be filtered between network subnets


Network with multiple Pulse systems

In scenarios where multiple Pulse systems are installed with dedicated servers on different subnets, IT administrator should block all multicast traffic between different Pulse systems. Still, Pulse handles such scenarios in the following way:

  1. Group calls: Each Pulse client will receive "Pulse server ID" during registration process and it will filter out all the group calls multicast traffic if it does not belong to this "server ID". In addition, each server will generate a set of multicast audio addresses based on it's unique MAC address.
  2. Station Discovery: During configuration of Pulse clients on sever StationWeb, discovery of all IP station on network is done. Those stations that are configured with another server IP address will automatically be deselected for configuration.


Server Call Forwarding

The Pulse Server supports server side call forwarding. When a station activates call forwarding it will notify the Pulse Server, which will store the call forwarding state. The Pulse Server will handle call forwarding for the station, thus the call forwarding will work even if the station is off-line. Forwarding status can be observed on the web page of the Pulse Server ("Server Monitoring"), where the target station of the forwarding is shown.

Station 204 is forwarded to station 203



Pulse trunking

Pulse Trunking is a feature in the Pulse Server for enabling calls between different Pulse sites. The function is enabled with Pulse-Server-Enterprise license. A total of 50 trunk routes can be configured on the Pulse server.

Pulse Trunk Routes must be configured to enable calls between different Pulse sites. A route consists of:

  1. Number range for the site the route goes to
  2. Logical name for the route
  3. IP address and SIP port of the remote Pulse Server on the remote site (Pulse Server SIP port is normally 5060)
  4. Profile to assign calls coming from/to the route. Profiles can be used to restrict access to/from a route

For two sites to call each other 2 routes must be configured. First, a route must be configured in Pulse Server 1, the route must define the number range of Pulse site 2 and the IP address of Pulse Server 2. In addition Pulse Server 2 must have a configured route to Pulse site 1, with the number range of Pulse site 1 and IP address of Pulse Server 1.

If only 1 route is enabled, site 1 with the Enterprise license can call site 2, but site 2 cannot call site 1

Example of Pulse Systems interconnected by Pulse Trunking


A client may only call to sites where there exists direct routes. In the example, the Remote sites may only call Central site, because only a route to Central site is configured. The Central site may call all of the Remote sites because it has routes configured to each Remote site. Configuration example for Central site Pulse server:

Configuration example of Pulse Trunking


Pulse SDK

Pulse system can be extended and customized with VS-SDK for Pulse. Key features are:

  • Build your own Vingtor-Stentofon Pulse intercom solution
  • Java and .NET APIs with ready-to-use examples and documentation (Intercom API)
  • Remotely control Turbine intercoms

SDK comes with easy to use API, out-of-box integration with Pulse Intercom system, support for all Turbine Intercom devices and PC soft-client, demo apps, code examples and API reference documentation. Intercom API even supports advanced Call center functions.

More info about VS-SDK for Pulse can be found in VS-SDK for Pulse (Intercom API).