Telephony

Latest revision as of 13:49, 27 October 2017

[edit] Session Initiation Protocol(SIP)

The Session Initiation Protocol(SIP) is an signaling protocol widely used for controlling communication sessions such as voice and video calls over Internet Protocol(IP).

About

Features

SIP in Yate

SIP protocol in Yate

SIP Security in Yate

TLS and SRTP in Yate

SIP Configuration File

Main configuration file for SIP module in Yate.

SIP Methods

How Yate processes SIP request methods and how to enable methods that are not handled by default.

SIP Features Module

SIP features module that implements SUBSCRIBE and NOTIFY Methods.

SIP Routing in Yate

Route to a SIP channel and a SIP Line.

SIP Client

Implementation and SIP Client features in Yate

SIP Features Module

SIP features module that implements SUBSCRIBE and NOTIFY methods.

SIP Send DTMFs

How to do configurations related to DTMFs in SIP channel.

SIP Attended Call Transfer In Cluster

How to configure Yate to handle a SIP attended call transfer for an unknown call leg in the cluster node.

SIP Flood Protection

Yate provides a protection mechanism against several types of SIP flood attacks.

SIP SBC

Describes how Yate can be used as a SIP session border controller.

xsip.generate

Use this message to initiate the transmission of a SIP request.

SIP query for CNAM and LNP

Feature used in a routing module that allows Yate to query CNAM and LNP databases over the SIP protocol.

SIP with NAT

Resolving SIP traversal problem by Yate.

How to use MESSAGE and PUBLISH requests

Example on how to handle custom SIP requests.

[edit] H323 Protocol

H.323 is a standard that specifies the components, protocols and procedures that provide multimedia communication services - real-time audio, video, and data communications - over packet-based networks.

About

Features

H323 channel module

Configuration file used to configure H323 protocol in Yate

Compiling Yate with OpenH323 support

Installing and compiling OpenH323 - stable version to use with Yate

Compiling Yate with H323plus support

How to compile Yate with H323plus support - unstable version to use with Yate

Yate as H323 GateKeeper and YateClient as H323 client

Configure H323 Gatekeeper And Multiple Endpoint Server

H323 To SIP Signalling Proxy

Configure in Yate H323 To SIP Signalling Proxy

H323 Send DTMFs

Describes DTMFs send related to h323 channel.

[edit] Media Gateway Control Protocol(MGCP)

Media Gateway Control Protocol is a signalling and call control protocol used within Voice over IP (VoIP) systems that typically inter-operate with the public switched telephone network (PSTN).

About

Features

MGCP call agent module

This module implements the Call Agent part of MGCP allowing Yate to control analog or digital Media Gateways

[edit] Inter-Asterisk eXchange(IAX/IAX2)

IAX2 is a VoIP protocol that carries both signaling and media on the same port.

About

Features

IAX

Main configuration file for IAX channel module in Yate.

Yate as IAX server and YateClient as IAX client

Setup to configure Yate as a IAX Server and YateClient as a IAX client

[edit] JINGLE Protocol

Jingle - Google talk/Google voice.

About

Features

JINGLE Module

How to configure JINGLE module in Yate

Jabber Client With Jingle Yate Server

How to configure a Jabber Client with Jingle Yate Server

Connecting to GMail

Learn how to route calls to contacts using Gmail account in accfile.conf

[edit] JABBER or XMPP(Extensible Messaging and Presence Protocol)

Extensible Messaging and Presence Protocol(XMPP) is a communications protocol for message-oriented middleware based on XML(Extensible Markup Language).

About

Features

Jabber Client Module

Configuring Yate as a Jabber Client

Jabber Server Module

Configuring Yate as a Jabber Server

Features for Jabber Server

The features for Jabber Server

Building Jabber server using Yate

Describes the steps to follow to build a Jabber server using Yate.

[edit] SS7 Protocol suite

Signaling System #7 (also known as C7) is a suite of communication protocols used in national and international telephony networks to provide signaling of calls and additional services.

Please read our Introduction to SS7 to get the basic concepts. This can help you find out what components exist in a SS7 network, how they interact and which ones you need.

About

Features

Introduction

SS7 Introduction

Terminology

Terminology for Yate implementation of SS7

SS7 Implementation

List on SS7 components inplemented in Yate.

Interconnecting on SS7

To connect to another SS7 network you will need to configure some parameters so the two sides speak the same language.

Signalling

This is a brief introduction on how the PSTN signalling modules are working together in Yate.

Usage Scenarios

Miscellaneous setups with various features: MTP2, MTP3, ISUP, CAMEL MAP, M3UA etc.

M2PA VS M2UA

A comparison between M2UA and M2PA

SCCP

SCCP represents a layer 4 protocol from SS7 signaling stack. It runs over a SS7 layer 3 protocol like MTP3 or M3UA

MTP2

MTP2 links are the classic way of interconnecting SS7 over TDM.

M3UA

This is a SIGTRAN protocol and it was designed for remote processing of SS7 messages above MTP3 protocol, mostly ISUP and SCCP.

M2PA

M2PA is a peer to peer SS7 layer 2 protocol. It was designed to replace MTP2 protocol in the IP scenarios.

M2UA

This is a SIGTRAN protocol allowing to access a remote MTP2 implementation on a Signaling Gateway.

[edit] Miscellaneous

Various articles that apply to more than one protocol

Articles

Call End Causes

Description of call end causes and errors in various protocols

See also

@@ Line 4: / Line 4: @@
 {|class="yate-table"
-|class="telephony_header_left"|'''About'''
+|class="yate-header-left"|'''About'''
-|class="telephony_header_right"|'''Features'''
+|class="yate-header-right"|'''Features'''
 |-
-|class="telephony-content-left"|
+|class="yate-content-left"|
 * [[SIP in Yate]]
 SIP protocol in Yate
@@ Line 23: / Line 23: @@
 Implementation and SIP Client features in Yate
-|class="telephony-content-right"|
+|class="yate-content-right"|
 * [[SIP Features Module]]
 SIP features module that implements SUBSCRIBE and NOTIFY methods.
@@ Line 49: / Line 49: @@
 {|class="yate-table"
-|class="telephony_header_left"|'''About'''
+|class="yate-header-left"|'''About'''
-|class="telephony_header_right"|'''Features'''
+|class="yate-header-right"|'''Features'''
 |-
-|class="telephony-content-left"|
+|class="yate-content-left"|
 * [[H323 channel module]]
 Configuration file used to configure H323 protocol in Yate
@@ Line 60: / Line 60: @@
 How to compile Yate with H323plus support - <font color="red">unstable version</font> to use with Yate
-|class="telephony-content-right"|
+|class="yate-content-right"|
 * [[Yate as H323 GateKeeper and YateClient as H323 client]]
 Configure H323 Gatekeeper And Multiple Endpoint Server
@@ Line 74: / Line 74: @@
 Media Gateway Control Protocol is a signalling and call control protocol used within Voice over IP (VoIP) systems that typically inter-operate with the public switched telephone network (PSTN).
-{|class="telephony"
+{|class="yate-table"
-|class="telephony_header_left"|'''About'''
+|class="yate-header-left"|'''About'''
-|class="telephony_header_right"|'''Features'''
+|class="yate-header-right"|'''Features'''
 |-
-|class="telephony-content-left"|
+|class="yate-content-left"|
-<div class="bullet">&nbsp;</div><font class="tel-title">[[MGCP call agent module]]</font><br/>
+* [[MGCP call agent module]]
-<div class="tel-desc">This module implements the Call Agent part of MGCP allowing Yate to control analog or digital Media Gateways</div>
+This module implements the Call Agent part of MGCP allowing Yate to control analog or digital Media Gateways
-|class="telephony-content-right"|
+|class="yate-content-right"|
 |}
@@ Line 87: / Line 88: @@
 IAX2 is a VoIP protocol that carries both signaling and media on the same port.
-{|class="telephony"
+{|class="yate-table"
-|class="telephony_header_left"|'''About'''
+|class="yate-header-left"|'''About'''
-|class="telephony_header_right"|'''Features'''
+|class="yate-header-right"|'''Features'''
 |-
-|class="telephony-content-left"|
+|class="yate-content-left"|
-<div class="bullet">&nbsp;</div><font class="tel-title">[[IAX]]</font><br/>
+* [[IAX]]
-<div class="tel-desc"> Main configuration file for IAX channel module in Yate.</div>
+Main configuration file for IAX channel module in Yate.
-|class="telephony-content-right"|
-<div class="bullet">&nbsp;</div><font class="tel-title">[[How to configure Yate as IAX server|Yate as IAX server and YateClient as IAX client]]</font><br/>
+|class="yate-content-right"|
-<div class="tel-desc">Setup to configure Yate as a IAX Server and YateClient as a IAX client</div>
+* [[How to configure Yate as IAX server|Yate as IAX server and YateClient as IAX client]]
+Setup to configure Yate as a IAX Server and YateClient as a IAX client
 |}
@@ Line 104: / Line 106: @@
 <!--Jingle is an extension to the Extensible Messaging and Presence Protocol (XMPP) which adds peer-to-peer (P2P) session control (signaling) for multimedia interactions such as in Voice over IP (VoIP) or videoconferencing communications.-->
-{|class="telephony"
+{|class="yate-table"
-|class="telephony_header_left"|'''About'''
+|class="yate-header-left"|'''About'''
-|class="telephony_header_right"|'''Features'''
+|class="yate-header-right"|'''Features'''
 |-
-|class="telephony-content-left"|
+|class="yate-content-left"|
-<div class="bullet">&nbsp;</div><font class="tel-title">[[Yjinglechan| JINGLE Module]]</font><br/>
+* [[Yjinglechan| JINGLE Module]]
-<div class="tel-desc">How to configure JINGLE module in Yate</div>
+How to configure JINGLE module in Yate
-|class="telephony-content-right"|
-<div class="bullet">&nbsp;</div><font class="tel-title">[[Jabber Client With Jingle Yate Server]]</font><br/>
+|class="yate-content-right"|
-<div class="tel-desc">How to configure a Jabber Client with Jingle Yate Server</div>
+* [[Jabber Client With Jingle Yate Server]]
-<div class="bullet">&nbsp;</div><font class="tel-title">[[Connecting to GMail]] </font><br/>
+How to configure a Jabber Client with Jingle Yate Server
-<div class="tel-desc">Learn how to route calls to contacts using Gmail account in accfile.conf</div>
+* [[Connecting to GMail]]
+Learn how to route calls to contacts using Gmail account in accfile.conf
 |}
@@ Line 122: / Line 125: @@
 Extensible Messaging and Presence Protocol(XMPP) is a communications protocol for message-oriented middleware based on XML(Extensible Markup Language).
-{|class="telephony"
+{|class="yate-table"
-|class="telephony_header_left"|'''About'''
+|class="yate-header-left"|'''About'''
-|class="telephony_header_right"|'''Features'''
+|class="yate-header-right"|'''Features'''
 |-
-|class="telephony-content-left"|
+|class="yate-content-left"|
-<div class="bullet">&nbsp;</div><font class="tel-title">[[Jabber Client Module]]</font><br/>
+* [[Jabber Client Module]]
-<div class="tel-desc">Configuring Yate as a Jabber Client</div>
+Configuring Yate as a Jabber Client
-<div class="bullet">&nbsp;</div><font class="tel-title">[[Jabber Server Module]]</font><br/>
+* [[Jabber Server Module]]
-<div class="tel-desc">Configuring Yate as a Jabber Server</div>
+Configuring Yate as a Jabber Server
-|class="telephony-content-right"|
-<div class="bullet">&nbsp;</div><font class="tel-title">[[Jbfeatures|Features for Jabber Server]]</font><br/>
+|class="yate-content-right"|
-<div class="tel-desc">The features for Jabber Server</div>
+* [[Jbfeatures|Features for Jabber Server]]
-<div class="bullet">&nbsp;</div><font class="tel-title">[[Building Jabber server using Yate]]</font><br/>
+The features for Jabber Server
-<div class="tel-desc">Describes the steps to follow to build a Jabber server using Yate.</div>
+* [[Building Jabber server using Yate]]
+Describes the steps to follow to build a Jabber server using Yate.
+|}
+==SS7 Protocol suite==
+[http://en.wikipedia.org/wiki/Signaling_System_7 Signaling System #7] (also known as C7) is a suite of communication protocols used in national and international telephony networks to provide signaling of calls and additional services.
+Please read our [[Introduction|Introduction to SS7]] to get the basic concepts. This can help you find out what components exist in a SS7 network, how they interact and which ones you need.
+{|class="yate-table"
+|class="yate-header-left"|'''About'''
+|class="yate-header-right"|'''Features'''
+|-
+|class="yate-content-left"|
+* [[Introduction]]
+SS7 Introduction
+* [[Terminology_for_Yate_implementation_of_SS7 | Terminology]]
+Terminology for Yate implementation of SS7
+* [[SS7 Implementation]]
+List on SS7 components inplemented in Yate.
+* [[Interconnecting on SS7]]
+To connect to another SS7 network you will need to configure some parameters so the two sides speak the same language.
+* [[Signalling]]
+This is a brief introduction on how the PSTN signalling modules are working together in Yate.
+|class="yate-content-right"|
+* [[How_To's#SS7_Setups | Usage Scenarios]]
+Miscellaneous setups with various features: MTP2, MTP3, ISUP, CAMEL MAP, M3UA etc.
+* [[M2PA VS M2UA]]
+A comparison between M2UA and M2PA
+* [[SCCP_Introduction | SCCP]]
+SCCP represents a layer 4 protocol from SS7 signaling stack. It runs over a SS7 layer 3 protocol like MTP3 or M3UA
+* [[SS7 MTP2 in Yate | MTP2]]
+MTP2 links are the classic way of interconnecting SS7 over TDM.
+* [[Configure M3UA | M3UA]]
+This is a SIGTRAN protocol and it was designed for remote processing of SS7 messages above MTP3 protocol, mostly ISUP and SCCP.
+* [[Configuring_M2PA | M2PA]]
+M2PA is a peer to peer SS7 layer 2 protocol. It was designed to replace MTP2 protocol in the IP scenarios.
+* [[Configure_M2UA | M2UA]]
+This is a SIGTRAN protocol allowing to access a remote MTP2 implementation on a Signaling Gateway.
 |}
@@ Line 145: / Line 188: @@
 |-
 |class="yate-content-right"|
-<div class="bullet">&nbsp;</div><font class="tel-title">[[Call End Causes]]</font><br/>
+* [[Call End Causes]]
-<div class="tel-desc">Description of call end causes and errors in various protocols</div>
+Description of call end causes and errors in various protocols
 |}
@@ Line 157: / Line 200: @@
 *[[About ISDN]]
 *[[Mobile_networks|Mobile Networks]]
+[[Category:SIP]] [[Category:H323]] [[Category:IAX]] [[Category:Jabber]] [[Category:Jingle]]

Telephony

Latest revision as of 13:49, 27 October 2017

Contents

[edit] Session Initiation Protocol(SIP)

[edit] H323 Protocol

[edit] Media Gateway Control Protocol(MGCP)

[edit] Inter-Asterisk eXchange(IAX/IAX2)

[edit] JINGLE Protocol

[edit] JABBER or XMPP(Extensible Messaging and Presence Protocol)

[edit] SS7 Protocol suite

[edit] Miscellaneous

Personal tools

Namespaces

Variants

Views

Actions

Search

Preface

Configuration

Administrators

Developers