• No results found

Session Initiation Protocol Capability Guide

N/A
N/A
Protected

Academic year: 2021

Share "Session Initiation Protocol Capability Guide"

Copied!
8
0
0

Loading.... (view fulltext now)

Full text

(1)

Session Initiation Protocol

Capability Guide

Introduction

The Session Initiation Protocol (SIP) driver provides Scout's

interface for VoIP telephony communications. The SIP driver

supports a variety of SIP servers in addition to Automatic Ring

Down (ARD) telephone lines. As with other Scout drivers, VPGate,

Scout's audio gateway, manages SIP connections.

Capabilities-at-a-Glance Scout's SIP implementation supports the following capabilities when they are available on the SIP server.

Capability Description

Controls

Patch A method for the console operator to connect calls to multiple other radio ortelephone endpoints. Activity History with

integrated Instant Recall Recorder

Tools that allow the console operator to review past conversations for analysis or clarification.

Transfer Transfer a call to a third party as either a blind transfer or an attended transfer. Automatic Ring Down A line that immediately dials a DTMF string to a specific endpoint when touched ifthe line is in idle or disconnect state. Hold Place a line on hold to stop the audio from being transmitted from or received at theScout console. Dial Tone Dial Tone button provides a dial tone for the endpoint; commonly referred to asRecall. Hookflash Flash button that simulates quickly hanging up then picking up again to send a hook-flash signal to the SIP server. Voicemail Visual indication on console alerts dispatcher that a voicemail is waiting.

(2)

Do Not Disturb

Activate Do Not Disturb on a console to block audible call indications for all endpoints on the console. All visual and audible call indications cease to endpoints configured as private when the console is in Do Not Disturb. Private endpoints reject calls or redirect calls to another endpoint.

Subscriber Signaling

Caller ID Subscriber Unit ID displayed on Scout screen on the endpoint pad and in ActivityHistory; Class caller ID via gateway or SIP signaling. Emergency Call Inbound emergency calls for an endpoint configured as an emergency line; all callsfrom the endpoint are emergency calls. DTMF Dual-Tone Multi-Frequency (DTMF) signaling, using 0 – 9, A – D, * and #.

High Availability Adding the High Availability driver to SIP endpoints allows for fast failover of anactive SIP call from one VPGate to another without dropping the call or losing a significant amount of audio.

(3)

SIP Connects Scout to Multiple Telephony Technologies

Scout simultaneously supports multiple VoIP telephony systems using SIP. VPGate registers extensions directly with the SIP Proxy Server and the Scout consoles access extensions through VPGate.

When using a SIP server with a Scout console, the console displays pads on the user interface that map to telephony circuits. These can be actual phone lines tied to a gateway or VoIP extensions off an IP PBX. Scout treats telephone circuits similarly to radios. A console can have multiple lines display on the screen and allow multiple phone calls to be active simultaneously.

In simple applications, 8-port SIP telephony gateways can be used to connect to POTS (plain old telephone service), foreign exchange office (FXO) lines, T1 lines, T1 PRI lines, E1 lines, foreign exchange station (FXS) lines, and Ear and Mouth (E&M) lines. For these scenarios, Scout supports connectivity using the following SIP telephony gateways.

Avtec Model Number Supports

(4)

GWC-8FXS-SIP Eight FXS Circuits GWC-E1-SIP One E1 Circuit GWC-8E/M-SIP Eight E&M Circuits

In a VoIP PBX-equipped installation, VPGate registers with the IP PBX and functions with the existing gateways (SIP servers) and desk telephones. Scout endpoints appear as extension registrations. Scout supports the following SIP servers for this scenario:

l Cisco Unified Communications Manager (formerly CallManager) Version 6.1 and higher l Avaya IP Office

l Generic SIP PBXs

High Availability Many customers who utilize SIP endpoints have a reliability expectation that is based on the reliability of the original telephone network. Scout emulates this reliability with its High Availability redundancy feature. VPGate provides redundancy at the endpoint level. Endpoints fail over based on theRedundant Priority

field value on the Endpoint Configuration webpage of the VPGate instances where the endpoints are configured. Adding theHigh Availability driver to SIP endpoints uses this same priority, but allows for fast

failover of an active SIP call from one VPGate to another without dropping the call or losing a significant amount of audio. Other than a slight loss of audio, the dispatcher should have no indication of the failover at the console.

NOTE

In order for a High Availability backup endpoint to recover an active SIP call, the other participant in the call must remain accessible when the original endpoint is lost. This prevents High Availability backup endpoints from recovering active calls between SIP endpoints that are on the same VPGate when that VPGate fails or loses network connectivity.

Qualified PBX Systems

Scout High Availability SIP endpoints have been qualified with Avaya and Cisco Call Manager. HA SIP endpoints are supported on Avaya utilizing SIP Trunking and SIP Extensions. HA SIP endpoints are

supported on Cisco Call Manager utilizing SIP Trunking. HA is not supported on Cisco Call Manager utilizing SIP Extensions.

Qualified PBX HA Support Utilizing SIP Trunking HA Support Utilizing SIP Extensions

Avaya Yes Yes

(5)
(6)

Licensing Avtec requires a Base VPGate license to interface with a SIP endpoint. Each SIP endpoint uses one Category B license slot in the base license. The following table shows the VPGate licenses available and the

maximum number of SIP endpoints (B licenses) for each.

NOTE

If configuring Scout's High Availability feature, a supplementary license is required for each SIP endpoint. See the VPGate Online Help for information regarding High Availability.

Avtec Model Number Maximum SIP Endpoints Redundant

SFW-VPG-L0-NR SFW-VPG-L0-NR-SK 12 No SFW-VPG-L0 SFW-VPG-L0-SK 12 Yes SFW-VPG-L1 SFW-VPG-L1-SK 20 Yes SFW-VPG-L2 SFW-VPG-L2-SK 40 Yes SFW-VPG-L3 SFW-VPG-L0-SK 100 Yes NOTE

Each of the VPGate licenses shown above also includes slots for Category A drivers. For more information on licensing, contact your Avtec sales representative.

Supported RFCs

(Request for Comments) Scout's SIP implementation is compliant with the following IETF Request for Comments (RFCs).

Core RFCs Title

RFC 2617 HTTP Authentication: Basic and Digest Access Authentication RFC 3261 SIP: Session Initiation Protocol

RFC 3263 Session Initiation Protocol (SIP): Locating SIP Servers

(7)

DNS-Related RFCs

RFC 3263 Session Initiation Protocol (SIP): Locating SIP Servers [also listed above] RFC 2181 Clarifications to the DNS Specification

RFC 2915 The Naming Authority Pointer (NAPTR) DNS Resource Record RFC 2782 A DNS RR for specifying the location of services (DNS SRV)

Extension RFCs RFC 2976 The SIP INFO Method

RFC 3265 Session Initiation Protocol (SIP)-Specific Event Notification RFC 3311 The Session Initiation Protocol (SIP) UPDATE Method

RFC 3326 The Reason Header Field for the Session Initiation Protocol (SIP) RFC 3420 Internet Media Type message/sipfrag

RFC 3515 The Session Initiation Protocol (SIP) Refer Method RFC 3891 The Session Initiation Protocol (SIP) "Replaces" Header RFC 3892 The Session Initiation Protocol (SIP) Referred-By Mechanism RFC 4028 Session Timers in the Session Initiation Protocol (SIP)

RFC 4488 Suppression of Session Initiation Protocol (SIP) REFER Method ImplicitSubscription RFC 4538 Request Authorization through Dialog Identification in the Session InitiationProtocol (SIP) RFC 5359 Session Initiation Protocol Service Examples

RFC 5589 Session Initiation Protocol (SIP) Call Control - Transfer RTP-Related RFCs

RFC 1889 Realtime Transport Protocol (RTP)

RFC 2833 / 4733 RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals RFC 3551 RTP Profile for Audio and Video Conferences with Minimal Control

(8)

liability resulting from errors or omissions in this document, or from the use of the information contained herein. Avtec, LLC reserves the right to make changes in the product design without reservation and without notification to its users. Avtec updates capability guides as changes occur. A capability guide could be the most current yet reflect a prior Scout release number if changes were not necessary at each release.

Scout™, VPGate™, Frontier™, Audio Bridge™, Avtec SIP Proxy™, ScoutLink™, CommScape™, and Outpost™ are trademarks of Avtec, LLC.

References

Related documents

The criteria are used to measure organizations’ struc- tures and functions that support patient-centered care concepts; human interactions; patient educa- tion and community access

Araştırmanın amacına bağlı olarak ele alınan ölçeklerin DFA aracılığı ile kuramsal yapı ile örtüştüğünün test edilmesiyle sosyal medya pazarlama

SIP allows additional features to be used, for example, sending a JPEG image and / or business card with the signaling – so that a called party can see who is calling.. Two

SIP Architecture Location Server Feature Server Registrar Server Proxy Server SIP Components Proxy Server. User Agent

If the Request-URI or top Route header field value contains a SIPS URI, the Contact header

Forking proxy example C sip.mci.com ACK INVITE INVITE 404 Not Found 180 Ringing INVITE sip:[email protected] host.wcom.com 180 Ringing ACK sip.uunet.com SIP User Agent Client

Free promo codes and game passes don’t work anymore so instead try some of the above methods to earn a nice amount of FREE ROBUX and make your gaming even..

Previous research of online gaming is briefly covered, along with literature that helps frame the virtual physical environment people experience when playing Runescape, and