A Convedia White Paper. Controlling Media Servers with SIP

Size: px
Start display at page:

Download "A Convedia White Paper. Controlling Media Servers with SIP"

Transcription

1 Version 1.2 June, 2004

2 Contents: Introduction page 3 Media Server Overview page 3 Dimensions of Interaction page 5 Types of Interaction page 6 SIP Standards for Media Server Control page 7

3 Introduction SIP is widely deployed today for VoIP network signaling and is enjoying great success. However, the use of SIP isn't limited to signaling. SIP has also been applied to the control of VoIP media servers, a function more traditionally associated with device control protocols such as MGCP or MEGACO. This article explores the kinds of media processing control that can be accomplished using SIP, the different control mechanisms that are available, suitable applications of each mechanism, and relevant standards. Media Server Overview A VoIP media server is a network element whose sole purpose is the processing of media streams, also known as RTP streams for network-based services. Media stream processing includes such functions as playing announcements, collecting DTMF digits, audio recording and playback, bridging multiple streams (also known as ), fax detection and decoding, speech recognition, text-to-speech rendering, and video processing. A VoIP media server is a network element whose sole purpose is the processing of media streams In performing these functions, the media server's role in the network is that of a slave device: it always operates under the direct control of one or more control agents, typically application servers and/or softswitches. A control agent provides a service execution environment, application-specific logic, and all the signaling for one or more services. The media server performs all the media processing for the service(s). This provides a division of responsibilities, or decomposition, between the control agent and the media server. The result of this decomposition is that the network signaling to and from VoIP gateways and terminals terminates on the control agent, while the media streams from the same gateways and terminals terminate on the media server. This separation of signaling (SIP) from media (RTP) is shown in Figure 1. Figure 1: Network Architecture 3 of 8

4 Decoupling signaling (SIP) and media (RTP) is not only elegant but also sensible Decoupling signaling (SIP) and media (RTP) is not only elegant but also sensible, given that service logic and signaling typically execute on network components based on generalpurpose processors, while carrier-class media processing is best achieved on network components based on Digital Signal Processors (DSPs). Control Agent / Media Server Interaction Although from a high-level point of view a control agent simply controls a media server, at a more detailed level there are actually two kinds of interactions: control messages and notification messages. Control messages move from the control agent to the media server, providing instructions for how to set up media streams and process them. Notification messages move from the media server to the control agent, providing results of the processing. In general control messages and notification messages occur asynchronously (that is, at any time) throughout the call. Table 1 shows some examples of control and notification messages between control agent (CA) and media server (MS) for some key media server features and services requiring those features. There are two kinds of interactions: control messages and notification messages MS Feature Announcements (1-way call) Example Application Network announcements Service announcements Control: CA to MS Audio file(s) to play None Notification: MS to CA Simple (N-way) Residential 3- or 6-way IP Centrex 3- or 6-way Call/conference to join None IVR in 1-way call Auto-attendant Card services Call screening Privacy services Audio file(s) to play or record DTMF digits or digit patterns to detect Grammars to recognize with ASR Text to speak with TTS (IVR script to execute) Detected digits Interruption by user Recognized words and phrases Fax detected or decoded IVR in 2-way call or small (3-way, 6-way, etc.) Prepaid card recharging in mid-call (2-way) Call center (2-way) Call center with supervisor monitoring/ coaching (3-way) Call/conference to join As for 1-way call above, but IVR is usually applied to a single leg of the conference Legs to mute/unmute As for 1-way call above Advanced (N-way) Voice Chat Business Event Conference to join As for 1-way call above, but IVR can be applied to one, several, or all legs Conference mixing mode (fixed speakers or loudest speakers) Legs to mute or change gain on As for 1-way call above Identities of current loudest speakers Table 1: Control Agent/Media Server Interaction 4 of 8

5 The simplest interactions are used in announcements and simple (the first two rows of the table), because these require no notifications, only a single control message at the start of the call. One-way IVR (the third row) is the next most complex, with both control and notification messages throughout a call but only a single party. The last two rows of the table show variants on what could be called "featured", that is, conference calls which differ from simple in using features such as announcements, IVR, muting, and mixing modes. These scenarios are the most complex, with a multi-party call having control and notification messages throughout the call on one or more legs. The main difference between these last two rows is the size of the conference, and what capabilities are therefore required. Examples of the use of these features include prepaid card recharging in mid-call (IVR in two-way call), supervisor interaction in call center calls (IVR in 3-way calls), and business or event which may range to thousands of participants (N-way calls). Dimensions of Interaction The information that needs to be exchanged between control agent and media server in the scenarios described above can be best understood by analyzing it along two independent dimensions. The first dimension of media server interaction is with respect to time. Examination of Table 1 shows that there are two of these types of interactions: those that occur at the start of a call and those that occur mid-call, that is, within a call. The first two rows, announcements and simple, need only start-of-call control messages. The last two rows, which involve multi-party calls, need mid-call messages (both control and notification), partly because of the need to control the multiple call legs. The third row, single-party IVR, also requires mid-call messages; however, it is often accomplished differently, using a start-of-call message followed by out-of-band (non-sip) messages for mid-call interaction. This leads to the second dimension of media server interaction: scripted versus unscripted interaction. VoiceXML is the best-known example of a voice services scripting language. The scope of VoiceXML is single-party IVR. It includes playing announcements, collecting DTMF digits, recording and playback, speech recognition, and text-to-speech. An IVR script may contain control logic that allows it to make decisions based on input and to perform various actions depending on those decisions. VoiceXML traditionally uses HTTP (not SIP) for interaction between the media server and control agent. This accounts for the mixed start-ofcall/mid-call character mentioned above: the script is carried to the media server at the start of the call using SIP, but subsequent mid-call interactions use HTTP. The first dimension of media server interaction is with respect to time The second dimension of media server interaction is scripted versus unscripted interaction Unscripted interactions, in contrast to the scripted case, make use of explicit start-of-call and/or mid-call messages between the control agent and the media server. The control agent sends a control message to the media server and the media server performs the requested action. This action can be immediate, such as playing an announcement, or can have a future effect, such as instructing the media server to report DTMF digits. In the latter example, the media server detects a digit at some point in the future, and sends a notification message to the control agent. 5 of 8

6 Types of Interaction The two dimensions of interaction lead naturally to four distinct possibilities for controlling a media server with SIP The two dimensions of interaction that have just been described, start-of-call versus mid-call and unscripted versus scripted, lead naturally to four distinct possibilities for controlling a media server with SIP. Figure 2 shows how the four combinations SIP alone, SIP and VoiceXML, SIP with mid-call interaction, and SIP with mid-call interaction and VoiceXML are derived from these two dimensions. As shown in Table 2, each of the four types of interactions has definite capabilities and limitations that make it most suitable for controlling a specific subset of a media server's features. Some of these features can be accomplished in only one way, while with others there is a choice between unscripted (using mid-call interaction) and scripted (using VoiceXML). Figure 2: Control Agent / Media Server Interaction Types MS Feature Only SIP SIP + VoiceXML SIP + Mid-Call SIP + Mid-Call + VoiceXML Announcements (1-way call) Simple (N-way) IVR in 1-way call IVR in 2-way call or small (3-way, 6-way, etc.) Advanced (N-way) Table 2: Interaction Types Suitability The simplest case, only SIP, can only be used with announcements and simple, as these basic interactions required only a start of call interaction. The next simplest case, "SIP and VoiceXML," supports single-party IVR using the VoiceXML scripting language. (VoiceXML script actually uses a simple type of mid-call interaction, HTTP GETs and POSTs, but this is very limited and not a full mid-call mechanism.) The third case, "SIP and mid-call", uses control and notification messages within a call to exchange information between the control agent and the media server. Table 2 shows that midcall interaction is an alternative to VoiceXML for single-party IVR calls. As shown in the last two rows in Table 2, the third case, "SIP and mid-call," and the fourth case, "SIP and mid-call and VoiceXML" are both possible alternatives for supporting a rich feature set in multi-party calls. Mid-call interactions can of course directly support IVR, using messages to play announcement and turn on digit collection and notification to report digits. However, they can also be used to invoke VoiceXML scripts at any time during a call or conference, which then interact out-of-band with the control agent using HTTP. Although this VoiceXML method can be applied to featured N-way (business or event), it is 6 of 8

7 most suitable for smaller multi-way calls that temporarily need IVR at some point, such as performing IVR in the middle of a two-party call. SIP Standards for Media Server Control The previous sections explored some technical aspects of media server control by SIP. This final section looks at the state of SIP standards for media server control. The two start-of-call cases, "only SIP" and "SIP and VoiceXML", which together handle announcements, simple, and single-party scripted IVR, are well understood in the industry and are described in an IETF draft ("draft-burger-sipping-netann") that has become a de facto standard. This public domain document shows how announcements, simple, and single-party scripted IVR are each invoked using a particular syntax of the Request-URI in the SIP INVITE message. The described Request-URI syntax is fully compliant with the SIP standard, and in no way modifies the operation of SIP itself. As a result, it is a straightforward matter for a softswitch or application server to make use of SIP to control a media server for these three functions. On the other hand, the two mid-call cases, "SIP and mid-call" and "SIP and mid-call and VoiceXML", are substantially more complex, and until very recently there was no suitable public domain standard for them. As Table 2 shows, for featured multi-party a mid-call interaction mechanism is a must-have, and featured multi-party cannot be supported with SIP alone or even with SIP/VoiceXML alone. Given that business and event is one of the leading VoIP applications today (and therefore an important driver for the VoIP industry) it is critical that the industry have access to SIP standards for mid-call media server interaction. The two start-of-call cases are well understood in the industry Until very recently, there was no suitable public domain standard for the two mid-call cases To address this, in June 2003 Convedia released two Internet Engineering Task Force (IETF) Internet Drafts that together define a powerful and extensible SIP-based control interface for IP media servers. The Drafts are called as follows: Media Sessions Markup Language (MSML) Media Objects Markup Language (MOML) MSML and MOML allow media servers to support functionality on SIP that previously could only be accomplished with MGCP, including capabilities dependent on mid-call interaction advanced and unscripted IVR. MSML and MOML are a simple addition to SIP and make use of XML content in the message bodies of SIP INVITE and INFO requests, without modifying the SIP protocol. MSML and MOML provide SIP-based standards for mid-call media server interaction Convedia believes that this new interface, which it is releasing into the public domain with no intellectual property rights (IPR) claims, is a significant step forward in the evolution of SIP media servers and will ultimately become the most widely used standard for media server control. 7 of 8

8 Copyright 2004 Convedia Corporation. All rights reserved. Convedia and the Convedia logo are trademarks of Convedia Corporation. All other trademarks are recognized as property of their respective owners. Information subject to change without notice.

White Paper Subcategory. Overview of XML Communication Technologies

White Paper Subcategory. Overview of XML Communication Technologies Subcategory Overview of XML Communication Technologies Executive Summary A significant shift has occurred in the communications infrastructures deployed today. This shift is the result of the acceptance

More information

Independent Submission Request for Comments: 5707 Category: Informational. Consultant February 2010

Independent Submission Request for Comments: 5707 Category: Informational. Consultant February 2010 Independent Submission Request for Comments: 5707 Category: Informational ISSN: 2070-1721 A. Saleem Y. Xin RadiSys G. Sharratt Consultant February 2010 Media Server Markup Language (MSML) Abstract The

More information

Phonologies The Voice of Technology

Phonologies The Voice of Technology Phonologies Media Services Framework Copyright 2004 Phonologies (India) Private Limited Copyright 2001 2004 by Phonologies (India) Private Limited. Phonologies, InterpreXer and Oktopous are trademarks

More information

Introducing the VoiceXML Server

Introducing the VoiceXML Server Introducing the VoiceXML Server David Asher Product Manager, Platform Solutions, NMS July 2005 Webinar Agenda Markets and introduction What is VoiceXML? System configurations Product description and features

More information

Back-end Avaya Aura Experience Portal and SIP-enabled Avaya Contact Center Select using a Play and Collect sample application

Back-end Avaya Aura Experience Portal and SIP-enabled Avaya Contact Center Select using a Play and Collect sample application Back-end Avaya Aura Experience Portal and SIP-enabled Avaya Contact Center Select using a Play and Collect sample application Overview This document describes how to integrate a back-end Avaya Aura Experience

More information

Back-end Avaya Aura Experience Portal and SIP-enabled Avaya Aura Contact Center using Context Creation

Back-end Avaya Aura Experience Portal and SIP-enabled Avaya Aura Contact Center using Context Creation Back-end Avaya Aura Experience Portal and SIP-enabled Avaya Aura Contact Center using Context Creation Overview This document describes how to integrate Avaya Aura Contact Center and a back-end Avaya Aura

More information

Authors Martin Eckert Ingmar Kliche Deutsche Telekom Laboratories.

Authors Martin Eckert Ingmar Kliche Deutsche Telekom Laboratories. Workshop on speaker biometrics and VoiceXML 3.0 March 5-6, 2009, Menlo Park, CA, US Proposal of an SIV architecture and requirements Authors Martin Eckert (martin.eckert@telekom.de), Ingmar Kliche (ingmar.kliche@telekom.de),

More information

Overview of SIP. Information About SIP. SIP Capabilities. This chapter provides an overview of the Session Initiation Protocol (SIP).

Overview of SIP. Information About SIP. SIP Capabilities. This chapter provides an overview of the Session Initiation Protocol (SIP). This chapter provides an overview of the Session Initiation Protocol (SIP). Information About SIP, page 1 How SIP Works, page 4 How SIP Works with a Proxy Server, page 5 How SIP Works with a Redirect Server,

More information

Application Notes for LumenVox Speech Engine with Avaya Voice Portal Issue 1.0

Application Notes for LumenVox Speech Engine with Avaya Voice Portal Issue 1.0 Avaya Solution & Interoperability Test Lab Application Notes for LumenVox Speech Engine with Avaya Voice Portal Issue 1.0 Abstract These Application Notes describe the configuration steps required to integrate

More information

Dialogic PowerMedia IP Media Server

Dialogic PowerMedia IP Media Server The Dialogic PowerMedia IP Media Server is a robust softwarebased multimedia server that allows service providers to rapidly deliver cost-effective video solutions with a high-quality user experience over

More information

Network Working Group Request for Comments: 4597 Category: Informational Cisco Systems, Inc. July 2006

Network Working Group Request for Comments: 4597 Category: Informational Cisco Systems, Inc. July 2006 Network Working Group Request for Comments: 4597 Category: Informational R. Even Polycom N. Ismail Cisco Systems, Inc. July 2006 Conferencing Scenarios Status of This Memo This memo provides information

More information

Network Working Group Request for Comments: 5167 Category: Informational Polycom March 2008

Network Working Group Request for Comments: 5167 Category: Informational Polycom March 2008 Network Working Group Request for Comments: 5167 Category: Informational M. Dolly AT&T Labs R. Even Polycom March 2008 Status of This Memo Media Server Control Protocol Requirements This memo provides

More information

VoiceXML. Installation and Configuration Guide. Interactive Intelligence Customer Interaction Center (CIC) Version 2016 R4

VoiceXML. Installation and Configuration Guide. Interactive Intelligence Customer Interaction Center (CIC) Version 2016 R4 VoiceXML Installation and Configuration Guide Interactive Intelligence Customer Interaction Center (CIC) Version 2016 R4 Last updated June 17, 2016 (See Change Log for summary of changes.) Abstract This

More information

IMS and Media Control. James Rafferty, Cantata Technology August 10, 2007

IMS and Media Control. James Rafferty, Cantata Technology August 10, 2007 IMS and Media Control James Rafferty, Cantata Technology August 10, 2007 IMS and Media Control IMS and Media Control Relationship to Overall IMS Media Resource Function Prior Art Relationship to Application

More information

A NOVEL MECHANISM FOR MEDIA RESOURCE CONTROL IN SIP MOBILE NETWORKS

A NOVEL MECHANISM FOR MEDIA RESOURCE CONTROL IN SIP MOBILE NETWORKS A NOVEL MECHANISM FOR MEDIA RESOURCE CONTROL IN SIP MOBILE NETWORKS Noël CRESPI, Youssef CHADLI, Institut National des Telecommunications 9, rue Charles Fourier 91011 EVRY Cedex FRANCE Authors: N.Crespi,

More information

Advanced VoIP Applications

Advanced VoIP Applications Advanced VoIP Applications New application deployments for VoIP networks can use a variety of network protocols and architectures. The use of MGCP and SIP are possible solutions and this paper discusses

More information

Media Resource Control Protocol v2

Media Resource Control Protocol v2 Media Resource Control Protocol v2 Sarvi Shanmugham, Editor: MRCP v1/v2 Technical Leader, Cisco Systems Session Number 1 Roadmap Overview of the IETF Speechsc WG Effort MRCP Short Summary MRCP Architecture

More information

OKI ADPCM, linear A-law and µ-law PCM, and Wave

OKI ADPCM, linear A-law and µ-law PCM, and Wave Intel NetStructure Host Media Processing Software Release 1.1 for the Windows* Operating System Media Processing Software for Building Cost-Effective IP Media Servers Features Implemented as a software-only

More information

GVP Deployment Guide. How the Media Control Platform Works

GVP Deployment Guide. How the Media Control Platform Works GVP Deployment Guide How the Media Control Platform Works 7/23/2018 How the Media Control Platform Works Read here about how the Media Control Platform performs its role in a GVP deployment: Operational

More information

MRCP Version 1. A.1 Overview

MRCP Version 1. A.1 Overview A MRCP Version 1 MRCP Version 1 (MRCPv1) is the predecessor to the MRCPv2 protocol. MRCPv1 was developed jointly by Cisco, Nuance and Speechworks, and is published under RFC 4463 [13]. MRCPv1 is an Informational

More information

VClarity Voice Platform

VClarity Voice Platform VClarity Voice Platform VClarity L.L.C. Voice Platform Snap-in Functional Overview White Paper Technical Pre-release Version 2.0 for VClarity Voice Platform Updated February 12, 2007 Table of Contents

More information

Application Notes for Nuance OpenSpeech Attendant with Avaya Voice Portal Issue 1.0

Application Notes for Nuance OpenSpeech Attendant with Avaya Voice Portal Issue 1.0 Avaya Solution & Interoperability Test Lab Application Notes for Nuance OpenSpeech Attendant with Avaya Voice Portal Issue 1.0 Abstract These Application Notes describe the configuration steps required

More information

Session Initiation Protocol (SIP)

Session Initiation Protocol (SIP) Session Initiation Protocol (SIP) Introduction A powerful alternative to H.323 More flexible, simpler Easier to implement Advanced features Better suited to the support of intelligent user devices A part

More information

Application Notes for Telisma telispeech Automatic Speech Recognition Engine with Avaya Voice Portal - Issue 1.0

Application Notes for Telisma telispeech Automatic Speech Recognition Engine with Avaya Voice Portal - Issue 1.0 Avaya Solution & Interoperability Test Lab Application Notes for Telisma telispeech Automatic Speech Recognition Engine with Avaya Voice Portal - Issue 1.0 Abstract These Application Notes describe the

More information

Abstract. Avaya Solution & Interoperability Test Lab

Abstract. Avaya Solution & Interoperability Test Lab Avaya Solution & Interoperability Test Lab Application Notes for Interactions Curo Speech Automated Speech Recognizer and Text-to-Speech Server with Avaya Aura Experience Portal using MRCP V2 Issue 1.0

More information

Application Notes for Yandex Speechkit Speech Recognition 1.6 with Avaya Aura Experience Portal Issue 1.0

Application Notes for Yandex Speechkit Speech Recognition 1.6 with Avaya Aura Experience Portal Issue 1.0 Avaya Solution & Interoperability Test Lab Application Notes for Yandex Speechkit Speech Recognition 1.6 with Avaya Aura Experience Portal 7.0.1 - Issue 1.0 Abstract These application notes describe the

More information

VoIP Basics. 2005, NETSETRA Corporation Ltd. All rights reserved.

VoIP Basics. 2005, NETSETRA Corporation Ltd. All rights reserved. VoIP Basics Phone Network Typical SS7 Network Architecture What is VoIP? (or IP Telephony) Voice over IP (VoIP) is the transmission of digitized telephone calls over a packet switched data network (like

More information

Gateway Options. PSTN Gateway, page 2

Gateway Options. PSTN Gateway, page 2 Cisco offers a large range of voice gateway models to cover a large range of requirements. Many, but not all, of these gateways have been qualified for use with Unified CVP. For the list of currently supported

More information

Dialogic PowerMedia Host Media Processing Software Release 3.0Win

Dialogic PowerMedia Host Media Processing Software Release 3.0Win Dialogic PowerMedia Host Media Processing Software Release 3.0Win (PowerMedia HMP 3.0) extends the capabilities of software-based IP media processing by introducing security features, video messaging,

More information

Dialogic PowerVille CC Cloud Centrex

Dialogic PowerVille CC Cloud Centrex Dialogic PowerVille CC Cloud Centrex Cloud-based Feature-rich Integrated VoIP Solution for Business and Residential Customers Dialogic s PowerVille Cloud Centrex is a carrier-class solution that enables

More information

Dialogic Cloud Centrex

Dialogic Cloud Centrex Dialogic Cloud Centrex Cloud-based, feature-rich integrated VoIP solution for business and residential customers Dialogic Cloud Centrex is a carrier-class solution that enables service providers to offer

More information

IP Multimedia Subsystem Application Servers

IP Multimedia Subsystem Application Servers IP Multimedia Subsystem Application Servers Second part of the project Presented by: Masood Khosroshahy B E G I N N I N G 1 June 2006 Project supervisor: Prof. Elie Najm IMS Application Servers HSS IMS

More information

Rev

Rev Rev. 2.8.1 Copyright Notice Copyright 2010-2017 Telinta Inc. No part of this document may be reproduced or transmitted in any form or by any means, electronic or mechanical, for any purpose, without the

More information

Collaborate App for Android Tablets

Collaborate App for Android Tablets The AT&T Collaborate service provides the Collaborate app to help you manage calls and conferences on your Android tablet on the go. The Collaborate app for Android tablets provides these communication

More information

ETSI TS V ( )

ETSI TS V ( ) TS 123 333 V10.3.0 (2012-01) Technical Specification Universal Mobile Telecommunications System (UMTS); LTE; Multimedia Resource Function Controller (MRFC) - Multimedia Resource Function Processor (MRFP)

More information

Mid-call Signaling Consumption

Mid-call Signaling Consumption The Cisco Unified Border Element BE Mid-call Signaling support aims to reduce the interoperability issues that arise due to consuming mid-call RE-INVITES/UPDATES. Mid-call Re-INVITEs/UPDATEs can be consumed

More information

SCOPIA Elite 5000 Series MCU

SCOPIA Elite 5000 Series MCU SCOPIA Elite 5000 Series MCU User Guide Version 7.7 2000-2011 RADVISION Ltd. All intellectual property rights in this publication are owned by RADVISION Ltd and are protected by United States copyright

More information

Application Notes for Configuring Computer Instruments Experience Configuration Interface, with Avaya Aura Experience Portal Issue 1.

Application Notes for Configuring Computer Instruments Experience Configuration Interface, with Avaya Aura Experience Portal Issue 1. Avaya Solution & Interoperability Test Lab Application Notes for Configuring Computer Instruments Experience Configuration Interface, with Avaya Aura Experience Portal Issue 1.0 Abstract These Application

More information

Abstract. Avaya Solution & Interoperability Test Lab

Abstract. Avaya Solution & Interoperability Test Lab Avaya Solution & Interoperability Test Lab Application Notes for LumenVox Automated Speech Recognizer, LumenVox Text-to-Speech Server and Call Progress Analysis with Avaya Aura Experience Portal Issue

More information

Obr.: a. SI2000/SI3000 ics integrated Call Server

Obr.: a. SI2000/SI3000 ics integrated Call Server SI2000/SI3000 integrated Call Server Preface The deregulation and removal of monopolies in the telecommunications market and the arrival of new technologies has enabled service providers to reduce their

More information

Application Notes for NMS Communications Vision Media Gateway Model VG2000 with Avaya Voice Portal and Avaya SIP Enablement Services Issue 1.

Application Notes for NMS Communications Vision Media Gateway Model VG2000 with Avaya Voice Portal and Avaya SIP Enablement Services Issue 1. Avaya Solution & Interoperability Test Lab Application Notes for NMS Communications Vision Media Gateway Model VG2000 with Avaya Voice Portal and Avaya SIP Enablement Services Issue 1.0 Abstract These

More information

Unified CVP Architecture Overview

Unified CVP Architecture Overview CHAPTER 1 Over the past two decades, many customers have invested in TDM-based interactive voice response (IVR) applications to automate simple customer transactions such as checking account or 401K account

More information

ETSI TS V7.3.1 ( ) Technical Specification

ETSI TS V7.3.1 ( ) Technical Specification TS 123 333 V7.3.1 (2008-04) Technical Specification Universal Mobile Telecommunications System (UMTS); Multimedia Resource Function Controller () - Multimedia Resource Function Processor () Mp interface;

More information

Multi-Service Access and Next Generation Voice Service

Multi-Service Access and Next Generation Voice Service Hands-On Multi-Service Access and Next Generation Voice Service Course Description The next generation of telecommunications networks is being deployed using VoIP technology and soft switching replacing

More information

WebEx Audio. Features

WebEx Audio. Features WebEx Integrated Audio provides a high-performance, feature-rich, telephony-based audio conference service. This service can be used in a stand-alone mode or fully integrated within a WebEx meeting. s,

More information

Configuring SIP Support for Hookflash

Configuring SIP Support for Hookflash Configuring SIP Support for Hookflash Last Updated: September 28, 2012 This chapter contains information about the SIP Support for Hookflash feature that allows you to configure IP Centrex supplementary

More information

Advanced and Customized Net Conference With Cisco WebEx Meeting Center Participant Quick Tips

Advanced and Customized Net Conference With Cisco WebEx Meeting Center Participant Quick Tips Advanced and Customized Net Conference With Cisco WebEx Meeting Center Participant Quick Tips Participant Quick Tips for WebEx Meeting Center provides tips that you can use to effectively join and participate

More information

Gateway Options. PSTN Gateway. PSTN Gateway, page 1

Gateway Options. PSTN Gateway. PSTN Gateway, page 1 PSTN Gateway, page 1 VoiceXML Gateway with or ASR/TTS, page 2 PSTN Gateway with or ASR/TTS, page 2 TDM Interfaces, page 2 Cisco Unified Border Element, page 3 Mixed G.729 and G.711 Codec Support, page

More information

2FXS Analog Telephone Adapter

2FXS Analog Telephone Adapter 2FXS Analog Telephone Adapter Product features Feature-rich telephone service over home or office Internet/ Intranet connection Auto-provisioning features for flexible, ease-of use IP PBX system integration

More information

CertifyMe. CertifyMe

CertifyMe. CertifyMe CertifyMe Number: 642-241 Passing Score: 800 Time Limit: 120 min File Version: 9.6 http://www.gratisexam.com/ CertifyMe 642-241 Exam A QUESTION 1 In a Cisco Unified Contact Center Enterprise design, the

More information

AT&T Collaborate glossary

AT&T Collaborate glossary Common terms associated with the AT&T Collaborate SM service. A B C D E F G H I J K L M N O P Q R S T U V W X Y Z A account codes A feature that lets administrators track and manage outgoing calls to keep

More information

Troubleshooting Voice Over IP with WireShark

Troubleshooting Voice Over IP with WireShark Hands-On Troubleshooting Voice Over IP with WireShark Course Description Voice over IP is being widely implemented both within companies and across the Internet. The key problems with IP voice services

More information

BT SIP Trunk Configuration Guide

BT SIP Trunk Configuration Guide CUCM 9.1 BT SIP Trunk Configuration Guide This document covers service specific configuration required for interoperability with the BT SIP Trunk service. Anything which could be considered as normal CUCM

More information

IPNext 187 Hybrid IP-PBX System High-performance Hybrid IP-PBX Solution

IPNext 187 Hybrid IP-PBX System High-performance Hybrid IP-PBX Solution IPNext 187 Hybrid IP-PBX System High-performance Hybrid IP-PBX Solution IP-PBX Features www.addpac.com AddPac Technology 2011, Sales and Marketing Contents IP-PBX Features Smart Multimedia Manager VoIP

More information

Application Notes for Anhui USTC iflytek InterReco with Avaya Aura Experience Portal Issue 1.0

Application Notes for Anhui USTC iflytek InterReco with Avaya Aura Experience Portal Issue 1.0 Avaya Solution & Interoperability Test Lab Application Notes for Anhui USTC iflytek InterReco with Avaya Aura Experience Portal Issue 1.0 Abstract These Application Notes describe the configuration steps

More information

Remote Support. User Guide 7.23

Remote Support. User Guide 7.23 Remote Support User Guide 7.23 Copyright 1997 2011 Cisco and/or its affiliates. All rights reserved. WEBEX, CISCO, Cisco WebEx, the CISCO logo, and the Cisco WebEx logo are trademarks or registered trademarks

More information

WebEx Participant Guide

WebEx Participant Guide WebEx Participant Guide Tufts Technology Services Training and Documentation WebEx Participant Guide 1 Table of Contents An Introduction to WebEx... 3 What is WebEx?... 3 Do I Need to Install Software

More information

Expandable SIP Phone System. Expandable SIP Phone System

Expandable SIP Phone System. Expandable SIP Phone System Expandable SIP Phone System Key Features Included: + One DVX-1000 SIP IP PBX + One DIV-140 Trunk Gateway + Ten DPH-140S IP Telephones + Unified Management + Save On Long-distance Calling + Create an IP

More information

SERIES Q: SWITCHING AND SIGNALLING Signalling requirements and protocols for the NGN Service and session control protocols supplementary services

SERIES Q: SWITCHING AND SIGNALLING Signalling requirements and protocols for the NGN Service and session control protocols supplementary services International Telecommunication Union ITU-T Q.3613 TELECOMMUNICATION STANDARDIZATION SECTOR OF ITU (05/2012) SERIES Q: SWITCHING AND SIGNALLING Signalling requirements and protocols for the NGN Service

More information

Category: Informational June An Architectural Framework for Media Server Control

Category: Informational June An Architectural Framework for Media Server Control Network Working Group T. Melanchuk, Ed. Request for Comments: 5567 Rain Willow Communications Category: Informational June 2009 Status of This Memo An Architectural Framework for Media Server Control This

More information

Overview of the Session Initiation Protocol

Overview of the Session Initiation Protocol CHAPTER 1 This chapter provides an overview of SIP. It includes the following sections: Introduction to SIP, page 1-1 Components of SIP, page 1-2 How SIP Works, page 1-3 SIP Versus H.323, page 1-8 Introduction

More information

Category: Standards Track October 2009

Category: Standards Track October 2009 Network Working Group J. Rosenberg Request for Comments: 5629 Cisco Systems Category: Standards Track October 2009 Abstract A Framework for Application Interaction in the Session Initiation Protocol (SIP)

More information

Advanced and Customized Net Conference With Cisco WebEx Meeting Center Participant Quick Tips

Advanced and Customized Net Conference With Cisco WebEx Meeting Center Participant Quick Tips Advanced and Customized Net Conference With Cisco WebEx Meeting Center Participant Quick Tips Participant Quick Tips for WebEx Meeting Center provides tips that you can use to effectively join and participate

More information

Oracle Communications Interactive Session Recorder and Broadsoft Broadworks Interoperability Testing. Technical Application Note

Oracle Communications Interactive Session Recorder and Broadsoft Broadworks Interoperability Testing. Technical Application Note Oracle Communications Interactive Session Recorder and Broadsoft Broadworks Interoperability Testing Technical Application Note Disclaimer The following is intended to outline our general product direction.

More information

Configure Nuance TTS and ASR for Cisco Unified Contact Center Enterprise (UCCE)

Configure Nuance TTS and ASR for Cisco Unified Contact Center Enterprise (UCCE) Configure Nuance TTS and ASR for Cisco Unified Contact Center Enterprise (UCCE) Contents Introduction Prerequisites Requirements Components Used Basic Configuration Cisco UCCE Configuration on VVB Configuration

More information

MGCP controls telephony gateways from a centralized call agent. This topic describes MGCP and identifies its associated standards.

MGCP controls telephony gateways from a centralized call agent. This topic describes MGCP and identifies its associated standards. Configuring MGCP MGCP and Its Associated Standards MGCP controls telephony gateways from a centralized call agent. This topic describes MGCP and identifies its associated standards. MGCP and Associated

More information

Dialogic PowerMedia HMP for Windows

Dialogic PowerMedia HMP for Windows Dialogic PowerMedia HMP for Windows Dialogic PowerMedia HMP for Windows (HMP Windows) is scalable, feature-rich media processing software for building innovative and costeffective voice solutions suitable

More information

Configuring SIP Call-Transfer Features

Configuring SIP Call-Transfer Features Configuring SIP Call-Transfer Features Configuring SIP Call-Transfer Features Last Updated: May 05, 2011 This chapter describes how to configure SIP call-transfer features. It describes the following features:

More information

Unified Customer Voice Portal Overview

Unified Customer Voice Portal Overview Overview, on page 1 Unified CVP Product Components, on page 2 Additional Components, on page 5 Call Flows, on page 14 Design Process, on page 16 Overview The Unified Customer Voice Portal (Unified CVP)

More information

Application Notes for OneAccess-Telstra Business SIP with Avaya IP Office Release 11 SIP Trunking - Issue 1.0

Application Notes for OneAccess-Telstra Business SIP with Avaya IP Office Release 11 SIP Trunking - Issue 1.0 Avaya Solution & Interoperability Test Lab Application Notes for OneAccess-Telstra Business SIP with Avaya IP Office Release 11 SIP Trunking - Issue 1.0 Abstract These Application Notes illustrate a sample

More information

quick start card Using AT&T Connect on Mac For participants, hosts and presenters

quick start card Using AT&T Connect on Mac For participants, hosts and presenters quick start card Using AT&T Connect on Mac For participants, hosts and presenters 2016 AT&T Intellectual Property. All rights reserved. AT&T, the AT&T logo and all other AT&T marks contained herein are

More information

Application Notes for Beijing InfoQuick SinoVoice Speech Technology (SinoVoice) jtts with Avaya Interactive Response Issue 1.0

Application Notes for Beijing InfoQuick SinoVoice Speech Technology (SinoVoice) jtts with Avaya Interactive Response Issue 1.0 Avaya Solution & Interoperability Test Lab Application Notes for Beijing InfoQuick SinoVoice Speech Technology (SinoVoice) jtts with Avaya Interactive Response Issue 1.0 Abstract These Application Notes

More information

Unified CVP Call Flow Models

Unified CVP Call Flow Models After understanding the Prerequisites for Call Flow Model Configuration, select one of the following call flow models for Unified Customer Voice Portal (CVP) implementation. Common Tasks for, page 1 Standalone

More information

VoIP Core Technologies. Aarti Iyengar Apricot 2004

VoIP Core Technologies. Aarti Iyengar Apricot 2004 VoIP Core Technologies Aarti Iyengar Apricot 2004 Copyright 2004 Table Of Contents What is Internet Telephony or Voice over IP? VoIP Network Paradigms Key VoIP Protocols Call Control and Signaling protocols

More information

Meet-Me Conferencing Quick Reference Guide MEET-ME CONFERENCING. Create Conferences

Meet-Me Conferencing Quick Reference Guide MEET-ME CONFERENCING. Create Conferences MEET-ME CONFERENCING Create Conferences Access You have access to Conferencing features if you have been assigned to a Meet-Me conference bridge. 1. Log in to BroadWorks. 2. On the Options list, click

More information

TSIN02 - Internetworking

TSIN02 - Internetworking Lecture 8: SIP and H323 Litterature: 2004 Image Coding Group, Linköpings Universitet Lecture 8: SIP and H323 Goals: After this lecture you should Understand the basics of SIP and it's architecture Understand

More information

Using the Cisco Unified Videoconferencing 5000 MCU

Using the Cisco Unified Videoconferencing 5000 MCU 2 CHAPTER Using the Cisco Unified Videoconferencing 5000 MCU This section describes how to create, join and manage video conferences on the MCU. Cisco Unified Videoconferencing 5000 MCU Access Levels,

More information

Unified Customer Voice Portal Overview

Unified Customer Voice Portal Overview Overview, page 1 Unified CVP Product Components, page 2 Additional Components, page 5 Call Flows, page 13 Design Process, page 14 Overview The Unified Customer Voice Portal (Unified CVP) is a web-based

More information

Abstract. These Application Notes describe the procedures for configuring Computer Instruments eci to interoperate with Avaya Voice Portal.

Abstract. These Application Notes describe the procedures for configuring Computer Instruments eci to interoperate with Avaya Voice Portal. Avaya Solution & Interoperability Test Lab Application Notes for Configuring Computer Instruments Experience Configuration Integration VoiceXML Application (eci), with Avaya Voice Portal Issue 1.0 Abstract

More information

Cisco Unified Customer Voice Portal 9.0

Cisco Unified Customer Voice Portal 9.0 Data Sheet Cisco Unified Customer Voice Portal 9.0 Product Overview Cisco Unified Customer Voice Portal (Unified CVP) is an award-winning product that provides IP-based selfservice and call routing. It

More information

Configuring SIP Call-Transfer Features

Configuring SIP Call-Transfer Features This chapter describes how to configure SIP call-transfer features. It describes the following features: SIP - Call Transfer Using Refer Method SIP - Call Transfer Enhancements Using Refer Method SIP Transfer

More information

Speech Applications. How do they work?

Speech Applications. How do they work? Speech Applications How do they work? What is a VUI? What the user interacts with when using a speech application VUI Elements Prompts or System Messages Prerecorded or Synthesized Grammars Define the

More information

What is CSTA? CSTA Overview. Started by Tom Miller (Siemens), updated by Ecma/TC32-TG11, December 2005.

What is CSTA? CSTA Overview. Started by Tom Miller (Siemens), updated by Ecma/TC32-TG11, December 2005. What is CSTA? CSTA Overview Started by Tom Miller (Siemens), updated by Ecma/TC32-TG11, December 2005. Topics CSTA History CSTA Standards Suite CSTA Features ECMA-323 (CSTA XML) Call Control Details Voice

More information

802.3af PoE SIP Analog Telephone Adapter

802.3af PoE SIP Analog Telephone Adapter 802.3af PoE SIP Analog Telephone Adapter Product features Feature-rich telephone service over home or office Internet/ Intranet connection 802.3af/at PoE and auto-provisioning features for flexible, ease-of

More information

SIP SIP Stack Portability

SIP SIP Stack Portability SIP SIP Stack Portability Implements capabilities to the SIP gateway Cisco IOS stack involving user-agent handling of messages, handling of unsolicited messages, support for outbound delayed media, and

More information

Cisco Cisco Voice over IP (CVOICE) Practice Test. Version QQ:

Cisco Cisco Voice over IP (CVOICE) Practice Test. Version QQ: Cisco 642-436 642-436 Cisco Voice over IP (CVOICE) Practice Test Version 3.8 QUESTION NO: 1 Cisco 642-436: Practice Exam Which two statements describe the purpose of the technology prefix? (Choose two.)

More information

Application Notes for Interactions Virtual Assistant Solutions with Avaya Aura Experience Portal Issue 1.0

Application Notes for Interactions Virtual Assistant Solutions with Avaya Aura Experience Portal Issue 1.0 Avaya Solution & Interoperability Test Lab Application Notes for Interactions Virtual Assistant Solutions with Avaya Aura Experience Portal Issue 1.0 Abstract These Application Notes describe the configuration

More information

Application Notes for the SDC IntelliSPEECH with Avaya Communication Manager - Issue 1.0

Application Notes for the SDC IntelliSPEECH with Avaya Communication Manager - Issue 1.0 Avaya Solution & Interoperability Test Lab Application Notes for the SDC IntelliSPEECH with Avaya Communication Manager - Issue 1.0 Abstract These Application Notes describe the configuration steps required

More information

Application Notes for VXi Connect Avaya Software and VXi Envoy UC USB Corded Headsets with Avaya one-x Communicator - Issue 1.0

Application Notes for VXi Connect Avaya Software and VXi Envoy UC USB Corded Headsets with Avaya one-x Communicator - Issue 1.0 Avaya Solution & Interoperability Test Lab Application Notes for VXi Connect Avaya Software and VXi Envoy UC USB Corded Headsets with Avaya one-x Communicator - Issue 1.0 Abstract These Application Notes

More information

A Technical Overview: Voiyager Dynamic Application Discovery

A Technical Overview: Voiyager Dynamic Application Discovery A Technical Overview: Voiyager Dynamic Application Discovery A brief look at the Voiyager architecture and how it provides the most comprehensive VoiceXML application testing and validation method available.

More information

You Only Have a Telephone?

You Only Have a Telephone? The Web Participant connects to the Web Conference using two channels: a communications channel from your computer to the host s AT&T Connect virtual meeting room and a sound channel from a telephone that

More information

Virtual Contact Center Implementation

Virtual Contact Center Implementation Virtual Contact Center Implementation JumpStart Training for the VCC Professional Plan Virtual Contact Center Implementation Please review this document to prepare for your JumpStart training sessions.

More information

Application Notes for Beijing InfoQuick SinoVoice Speech Technology (SinoVoice) jtts with Avaya Voice Portal Issue 1.0

Application Notes for Beijing InfoQuick SinoVoice Speech Technology (SinoVoice) jtts with Avaya Voice Portal Issue 1.0 Avaya Solution & Interoperability Test Lab Application Notes for Beijing InfoQuick SinoVoice Speech Technology (SinoVoice) jtts with Avaya Voice Portal Issue 1.0 Abstract These Application Notes describe

More information

Calabrio Recording Services. Deployment Guide for Cisco MediaSense

Calabrio Recording Services. Deployment Guide for Cisco MediaSense Calabrio Recording Services Deployment Guide for Cisco MediaSense Version First Published: September 30, 2014 Last Updated: June 3, 2014 Calabrio and Calabrio ONE are registered trademarks and the Calabrio

More information

SurVo. Stepping Through the Basics. Version 2.0

SurVo. Stepping Through the Basics. Version 2.0 Stepping Through the Basics Version 2.0 Contents What is a SurVo?... 3 SurVo: Voice Survey Form... 3 About the Documentation... 3 Ifbyphone on the Web... 3 Setting up a SurVo... 4 Speech/Recording Options...

More information

This is a sample chapter of WebRTC: APIs and RTCWEB Protocols of the HTML5 Real-Time Web by Alan B. Johnston and Daniel C. Burnett.

This is a sample chapter of WebRTC: APIs and RTCWEB Protocols of the HTML5 Real-Time Web by Alan B. Johnston and Daniel C. Burnett. This is a sample chapter of WebRTC: APIs and RTCWEB Protocols of the HTML5 Real-Time Web by Alan B. Johnston and Daniel C. Burnett. For more information or to buy the paperback or ebook editions, visit

More information

Deployment note. Products for conferencing platform developers Product deployment note

Deployment note. Products for conferencing platform developers Product deployment note Products for conferencing platform developers Product deployment note The Aculab solution The market needs The changing market conditions, lower cost of telephone calls, teleworking, business travel patterns

More information

Dialogic PowerMedia IP Media Server Release 3.1.0

Dialogic PowerMedia IP Media Server Release 3.1.0 Dialogic PowerMedia IP Media Server Release 3.1.0 Application Developer s Guide February 2011 64-0531-02 www.dialogic.com Copyright and Legal Notice Copyright 2000-2011 Dialogic Inc. All Rights Reserved.

More information

Big Capability For Small Business

Big Capability For Small Business STRATA CTX100 Big Capability For Small Business Communicate Better. It s time to break down the barriers to greater productivity, take a giant leap toward improved communications, and embrace one of today

More information