Telephony

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Yate is a telephony engine who's main function is to connect together call legs to external devices or to internal server resources. It uses a variety of VoIP protocols(as SIP, H.323, IAX2 or Jingle) that can be used without the need of special hardware. In the next chapters you will find more informations about configuring each telephony protocol and the features supported by Yate.
 
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==Session Initiation Protocol(SIP)==
 
==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).
 
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).
  
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{|class="telephony"
+
{|class="yate-table"
−
|class="telephony_header_left"|'''About'''
+
|class="yate-header-left"|'''About'''
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|class="telephony_header_right"|'''Features'''
+
|class="yate-header-right"|'''Features'''
 
|-
 
|-
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|class="telephony-content-left"|
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|class="yate-content-left"|
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<div class="bullet">&nbsp;</div><font class="tel-title">[[SIP in Yate]]</font><br/>
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* [[SIP in Yate]]
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<div class="tel-desc">SIP protocol in Yate</div>
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SIP protocol in Yate
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<div class="bullet">&nbsp;</div><font class="tel-title">[[SIP Security in Yate]]</font><br/>
+
* [[SIP Security in Yate]]
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<div class="tel-desc">TLS and SRTP in Yate</div>
+
TLS and SRTP in Yate
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<div class="bullet">&nbsp;</div><font class="tel-title">[[SIP Configuration File]]</font><br/>
+
* [[SIP Configuration File]]
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<div class="tel-desc">Main configuration file for SIP module in Yate.</div>
+
Main configuration file for SIP module in Yate.
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<div class="bullet">&nbsp;</div><font class="tel-title">[[SIP Methods]]</font><br/>
+
* [[SIP Methods]]
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<div class="tel-desc">How Yate processes SIP request methods and how to enable methods that are not handled by default.</div>
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How Yate processes SIP request methods and how to enable methods that are not handled by default.
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<div class="bullet">&nbsp;</div><font class="tel-title">[[SIP Features Module]]</font><br/>
+
* [[SIP Features Module]]
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<div class="tel-desc">SIP features module that implements SUBSCRIBE and NOTIFY Methods.</div>
+
SIP features module that implements SUBSCRIBE and NOTIFY Methods.
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<div class="bullet">&nbsp;</div><font class="tel-title">[[SIP Routing in Yate]]</font><br/> 
+
* [[SIP Routing in Yate]]
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<div class="tel-desc">Route to a SIP channel and a SIP Line.</div>
+
Route to a SIP channel and a SIP Line.
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<div class="bullet">&nbsp;</div><font class="tel-title">[[SIP Client]]</font><br/>
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* [[SIP Client]]
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<div class="tel-desc"> Implementation and SIP Client features in Yate </div>
+
Implementation and SIP Client features in Yate
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|class="telephony-content-right"|
+
 
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<div class="bullet">&nbsp;</div><font class="tel-title">[[SIP Features Module]]</font><br/>
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|class="yate-content-right"|
−
<div class="tel-desc">SIP features module that implements SUBSCRIBE and NOTIFY methods </div>
+
* [[SIP Features Module]]
−
<div class="bullet">&nbsp;</div><font class="tel-title">[[SIP Send DTMFs]]</font><br/>
+
SIP features module that implements SUBSCRIBE and NOTIFY methods.
−
<div class="tel-desc">How to do configurations related to DTMFs in SIP channel.</div>
+
* [[SIP Send DTMFs]]
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<div class="bullet">&nbsp;</div><font class="tel-title">[[SIP Attended Call Transfer In Cluster]]</font><br/>
+
How to do configurations related to DTMFs in SIP channel.
−
<div class="tel-desc">How to configure Yate to handle a SIP attended call transfer for an unknown call leg in the cluster node</div>
+
* [[SIP Attended Call Transfer In Cluster]]
−
<div class="bullet">&nbsp;</div><font class="tel-title">[[SIP Flood Protection]]</font><br/>
+
How to configure Yate to handle a SIP attended call transfer for an unknown call leg in the cluster node.
−
<div class="tel-desc">Yate provides a protection mechanism against several types of SIP flood attacks.</div>
+
* [[SIP Flood Protection]]
−
<div class="bullet">&nbsp;</div><font class="tel-title">[[SIP SBC]]</font><br/>
+
Yate provides a protection mechanism against several types of SIP flood attacks.
−
<div class="tel-desc">Describes how Yate can be used as a SIP session border controller.</div>
+
* [[SIP SBC]]
−
<div class="bullet">&nbsp;</div><font class="tel-title">[[xsip.generate]]</font><br/>
+
Describes how Yate can be used as a SIP session border controller.
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<div class="tel-desc">Use this message to initiate the transmission of a SIP request.</div>
+
* [[xsip.generate]]
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<div class="bullet">&nbsp;</div><font class="tel-title">[[SIP query for CNAM and LNP]]</font><br/>
+
Use this message to initiate the transmission of a SIP request.
−
<div class="tel-desc"> Feature used in a routing module that allows Yate to query CNAM and LNP databases over the SIP protocol.</div>
+
* [[SIP query for CNAM and LNP]]
−
<div class="bullet">&nbsp;</div><font class="tel-title">[[SIP NAT|SIP with NAT]]</font><br/>
+
Feature used in a routing module that allows Yate to query CNAM and LNP databases over the SIP protocol.
−
<div class="tel-desc">Resolving SIP traversal problem by Yate.</div><br/>
+
* [[SIP NAT|SIP with NAT]]
 +
Resolving SIP traversal problem by Yate.
 +
* [[How_to_setup_chat_and_short_file_transfer_using_MESSAGE_Request_Method|How to use MESSAGE and PUBLISH requests]]
 +
Example on how to handle custom SIP requests.
 
|}
 
|}
  
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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.
 
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.
  
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{|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">[[H323 channel module]]</font><br/>
+
* [[H323 channel module]]
−
<div class="tel-desc">Configuration file used to configure H323 protocol in Yate</div>
+
Configuration file used to configure H323 protocol in Yate
−
<div class="bullet">&nbsp;</div><font class="tel-title">[[OpenH323|Compiling Yate with OpenH323 support]]</font><br/>
+
* [[OpenH323|Compiling Yate with OpenH323 support]]
−
<div class="tel-desc">Installing and compiling OpenH323 - stable version to use with Yate</div>
+
Installing and compiling OpenH323 - stable version to use with Yate
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<div class="bullet">&nbsp;</div><font class="tel-title">[[Compiling Yate with H323plus support]]</font><br/>
+
* [[Compiling Yate with H323plus support]]
−
<div class="tel-desc">How to compile Yate with H323plus support - unstable version to use with Yate</div>
+
How to compile Yate with H323plus support - <font color="red">unstable version</font> to use with Yate
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|class="telephony-content-right"|
+
 
−
<div class="bullet">&nbsp;</div><font class="tel-title">[[H323 Gatekeeper And Multiple Endpoint Server]]</font><br/>
+
|class="yate-content-right"|
−
<div class="tel-desc"> Configure H323 Gatekeeper And Multiple Endpoint Server</div>
+
* [[Yate as H323 GateKeeper and YateClient as H323 client]]
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<div class="bullet">&nbsp;</div><font class="tel-title">[[H323 To SIP Signalling Proxy]]</font><br/>
+
Configure H323 Gatekeeper And Multiple Endpoint Server
−
<div class="tel-desc">Configure in Yate H323 To SIP Signalling Proxy</div>
+
* [[H323 To SIP Signalling Proxy]]
−
<div class="bullet">&nbsp;</div><font class="tel-title">[[H323 Send DTMFs]]</font><br/>
+
Configure in Yate H323 To SIP Signalling Proxy
−
<div class="tel-desc">Describes DTMFs send related to h323 channel.</div>
+
* [[H323 Send DTMFs]]
 +
Describes DTMFs send related to h323 channel.
 +
 
 
|}
 
|}
  
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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).
 
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"|
+
 
−
<div class="bullet">&nbsp;</div><font class="tel-title">[[Configuring Mediant MGCP]]</font><br/>
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|class="yate-content-right"|
−
<div class="tel-desc">How to configure Yate with Mediant MGCP</div>
+
−
<div class="bullet">&nbsp;</div><font class="tel-title">[[Configuration for a Mediant MP-114 FXS analog gateway]]</font><br/>
+
−
<div class="tel-desc">How to configure Yate to use a Mediant MP-114 FXS analog gateway</div>
+
 
|}
 
|}
  
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IAX2 is a VoIP protocol that carries both signaling and media on the same port.
 
IAX2 is a VoIP protocol that carries both signaling and media on the same port.
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{|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]]
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<div class="tel-desc"> Explanation how the connection works with trunked packets</div>
+
Main configuration file for IAX channel module in Yate.
−
|class="telephony-content-right"|
+
 
−
<div class="bullet">&nbsp;</div><font class="tel-title">[[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
 
|}
 
|}
  
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<!--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.-->
 
<!--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
 
|}
 
|}
  
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Extensible Messaging and Presence Protocol(XMPP) is a communications protocol for message-oriented middleware based on XML(Extensible Markup Language).
 
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/>
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* [[Jabber Client Module]]
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<div class="tel-desc">Configuring Yate as a Jabber Client</div>
+
Configuring Yate as a Jabber Client
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<div class="bullet">&nbsp;</div><font class="tel-title">[[Jabber Server Module]]</font><br/>
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* [[Jabber Server Module]]
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<div class="tel-desc">Configuring Yate as a Jabber Server</div>
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Configuring Yate as a Jabber Server
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|class="telephony-content-right"|
+
 
−
<div class="bullet">&nbsp;</div><font class="tel-title">[[Jbfeatures|Features for Jabber Server]]</font><br/>
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|class="yate-content-right"|
−
<div class="tel-desc">The features for Jabber Server</div>
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* [[Jbfeatures|Features for Jabber Server]]
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<div class="bullet">&nbsp;</div><font class="tel-title">[[Building Jabber server using Yate]]</font><br/>
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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.
 +
|}
 +
 +
==Miscellaneous==
 +
Various articles that apply to more than one protocol
 +
 +
{|class="yate-table"
 +
|class="yate-header-right"|'''Articles'''
 +
|-
 +
|class="yate-content-right"|
 +
* [[Call End Causes]]
 +
Description of call end causes and errors in various protocols
 +
|}
 +
  
  
 
'''See also'''
 
'''See also'''
  
 +
*[[About VoIP]]
 +
*[[About PSTN]]
 +
*[[About ISDN]]
 
*[[Mobile_networks|Mobile Networks]]
 
*[[Mobile_networks|Mobile Networks]]
 +
 +
[[Category:SIP]] [[Category:H323]] [[Category:IAX]] [[Category:Jabber]] [[Category:Jingle]]

Latest revision as of 13:49, 27 October 2017

Contents

[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 protocol in Yate

TLS and SRTP in Yate

Main configuration file for SIP module in Yate.

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

SIP features module that implements SUBSCRIBE and NOTIFY Methods.

Route to a SIP channel and a SIP Line.

Implementation and SIP Client features in Yate

SIP features module that implements SUBSCRIBE and NOTIFY methods.

How to do configurations related to DTMFs in SIP channel.

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

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

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

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

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

Resolving SIP traversal problem by Yate.

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

Configuration file used to configure H323 protocol in Yate

Installing and compiling OpenH323 - stable version to use with Yate

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

Configure H323 Gatekeeper And Multiple Endpoint Server

Configure in Yate H323 To SIP Signalling Proxy

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

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

Main configuration file for IAX channel module in Yate.

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

[edit] JINGLE Protocol

Jingle - Google talk/Google voice.

About Features

How to configure JINGLE module in Yate

How to configure a Jabber Client with Jingle Yate Server

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

Configuring Yate as a Jabber Client

Configuring Yate as a Jabber Server

The features for Jabber Server

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

SS7 Introduction

Terminology for Yate implementation of SS7

List on SS7 components inplemented in Yate.

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

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

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

A comparison between M2UA and M2PA

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

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

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

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

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

Description of call end causes and errors in various protocols


See also

Retrieved from "https://docs.yate.ro/wiki/index.php?title=Telephony&oldid=8227"
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