Telephony

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{|class="wikitable"
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==Session Initiation Protocol(SIP)==
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|colspan="12" style="text-align:center;font-weight:bold;" |Telephony modules in Yate
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 +
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).
 +
 
 +
{|class="yate-table"
 +
|class="yate-header-left"|'''About'''
 +
|class="yate-header-right"|'''Features'''
 
|-
 
|-
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|colspan="2" style="text-align:center;"|SIP
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|class="yate-content-left"|
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|colspan="2" style="text-align:center;"|H323
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* [[SIP in Yate]]
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|colspan="2" style="text-align:center;"|MGCP
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SIP protocol in Yate
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|colspan="2" style="text-align:center;"|IAX2(IAX1)
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* [[SIP Security in Yate]]
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|colspan="2" style="text-align:center;"|JINGLE
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TLS and SRTP in Yate
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|colspan="2" style="text-align:center;"|JABBER
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* [[SIP Configuration File]]
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Main configuration file for SIP module in Yate.
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* [[SIP Methods]]
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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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* [[SIP Features Module]]
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SIP features module that implements SUBSCRIBE and NOTIFY Methods.
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* [[SIP Routing in Yate]]
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Route to a SIP channel and a SIP Line.
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* [[SIP Client]]
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Implementation and SIP Client features in Yate
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|class="yate-content-right"|
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* [[SIP Features Module]]
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SIP features module that implements SUBSCRIBE and NOTIFY methods.
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* [[SIP Send DTMFs]]
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How to do configurations related to DTMFs in SIP channel.
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* [[SIP Attended Call Transfer In Cluster]]
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How to configure Yate to handle a SIP attended call transfer for an unknown call leg in the cluster node.
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* [[SIP Flood Protection]]
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Yate provides a protection mechanism against several types of SIP flood attacks.
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* [[SIP SBC]]
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Describes how Yate can be used as a SIP session border controller.
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* [[xsip.generate]]
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Use this message to initiate the transmission of a SIP request.
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* [[SIP query for CNAM and LNP]]
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Feature used in a routing module that allows Yate to query CNAM and LNP databases over the SIP protocol.
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* [[SIP NAT|SIP with NAT]]
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Resolving SIP traversal problem by Yate.
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* [[How_to_setup_chat_and_short_file_transfer_using_MESSAGE_Request_Method|How to use MESSAGE and PUBLISH requests]]
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Example on how to handle custom SIP requests.
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|}
 +
 
 +
== H323 Protocol==
 +
 
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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.
 +
 
 +
{|class="yate-table"
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|class="yate-header-left"|'''About'''
 +
|class="yate-header-right"|'''Features'''
 
|-
 
|-
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|About SIP||SIP Features||About H323|| H323 Features||About MGCP||MGCP Features||About IAX||IAX Features||About JINGLE||JINGLE Features||About JABBER||JABBER Features
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|class="yate-content-left"|
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* [[H323 channel module]]
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Configuration file used to configure H323 protocol in Yate
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* [[OpenH323|Compiling Yate with OpenH323 support]]
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Installing and compiling OpenH323 - stable version to use with Yate
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* [[Compiling Yate with H323plus support]]
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How to compile Yate with H323plus support - <font color="red">unstable version</font> to use with Yate
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|class="yate-content-right"|
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* [[Yate as H323 GateKeeper and YateClient as H323 client]]
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Configure H323 Gatekeeper And Multiple Endpoint Server
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* [[H323 To SIP Signalling Proxy]]
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Configure in Yate H323 To SIP Signalling Proxy
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* [[H323 Send DTMFs]]
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Describes DTMFs send related to h323 channel.
 +
 
 +
|}
 +
 
 +
== Media Gateway Control Protocol(MGCP) ==
 +
 
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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).
 +
 
 +
{|class="yate-table"
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|class="yate-header-left"|'''About'''
 +
|class="yate-header-right"|'''Features'''
 
|-
 
|-
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|[[SIP in Yate]]
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|class="yate-content-left"|
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|[[SIP Client]]
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* [[MGCP call agent module]]
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|[[H323 channel module]]
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This module implements the Call Agent part of MGCP allowing Yate to control analog or digital Media Gateways
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|[[H323 Gatekeeper And Multiple Endpoint Server]]
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|[[MGCP call agent module]]
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|class="yate-content-right"|
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|[[Configuring Mediant MGCP]]
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|}
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|[[IAX|IAX channel]]
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|[[Yate as IAX server and YateClient as IAX client]]
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== Inter-Asterisk eXchange(IAX/IAX2) ==
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|[[Yjinglechan]]
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|[[Jabber Client With Jingle Yate Server]]
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IAX2 is a VoIP protocol that carries both signaling and media on the same port.
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|[[Jabber Client Module]]
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{|class="yate-table"
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|[[Building Jabber server using Yate]]
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|class="yate-header-left"|'''About'''
 +
|class="yate-header-right"|'''Features'''
 
|-
 
|-
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| [[SIP Listeners in Yate]]
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|class="yate-content-left"|
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| [[SIP Features Module]]
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* [[IAX]]
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|
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Main configuration file for IAX channel module in Yate.
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|[[H323 To SIP Signalling Proxy]]
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−
|
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|class="yate-content-right"|
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|[[Configuration for a Mediant MP-114 FXS analog gateway]]
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* [[How to configure Yate as IAX server|Yate as IAX server and YateClient as IAX client]]
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|
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Setup to configure Yate as a IAX Server and YateClient as a IAX client
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|[[Connecting To IAXTEL]]
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|}
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|
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|[[Connecting to GMail]]
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== JINGLE Protocol==
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|[[Jabber Server Module]]
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|[[Jbfeatures|Jbfeatures Module]]
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Jingle - Google talk/Google voice.
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|-  
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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.-->
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|[[SIP Messages]]
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|[[SIP Send DTMFs]]
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{|class="yate-table"
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|
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|class="yate-header-left"|'''About'''
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|colspan="9"|[[H323 Send DTMFs]]
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|class="yate-header-right"|'''Features'''
 
|-
 
|-
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|[[SIP Routing in Yate]]
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|class="yate-content-left"|
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|[[SIP Attended Call Transfer In Cluster]]
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* [[Yjinglechan| JINGLE Module]]
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|
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How to configure JINGLE module in Yate
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|colspan="9"|[[Compiling Yate with H323plus support]]
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|-
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|class="yate-content-right"|
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|[[SIP Configuration File]]
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* [[Jabber Client With Jingle Yate Server]]
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|colspan="11"|[[SIP Flood Protection]]  
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How to configure a Jabber Client with Jingle Yate Server
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|-
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* [[Connecting to GMail]]
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|
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Learn how to route calls to contacts using Gmail account in accfile.conf
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|[[SIP Generic Message]]
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|}
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|colspan="10"
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|-
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==JABBER or XMPP(Extensible Messaging and Presence Protocol)==
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|
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|[[SIP SBC]]
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Extensible Messaging and Presence Protocol(XMPP) is a communications protocol for message-oriented middleware based on XML(Extensible Markup Language).
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|colspan="10"|
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−
|-
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{|class="yate-table"
−
|
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|class="yate-header-left"|'''About'''
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|[[xsip.generate]]
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|class="yate-header-right"|'''Features'''
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|colspan="10"|
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−
|-
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−
|
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|[[SIP query for CNAM and LNP]]
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−
|colspan="10"|
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−
|-
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−
|
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|[[SIP NAT]]
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|colspan="10"|
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|-
 
|-
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|class="yate-content-left"|
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* [[Jabber Client Module]]
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Configuring Yate as a Jabber Client
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* [[Jabber Server Module]]
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Configuring Yate as a Jabber Server
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 +
|class="yate-content-right"|
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* [[Jbfeatures|Features for Jabber Server]]
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The features for Jabber Server
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* [[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.
  
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{|class="yate-table"
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{|class="wikitable"
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|class="yate-header-left"|'''About'''
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!Telephony modules
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|class="yate-header-right"|'''Features'''
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! Features
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|-
 
|-
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|[[SIP channel module|SIP]]
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|class="yate-content-left"|
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|[[SIP features]]
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* [[Introduction]]
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|-
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SS7 Introduction
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|[[H323_channel_module|H323]]
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* [[Terminology_for_Yate_implementation_of_SS7 | Terminology]]
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|[[H323 features]]
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Terminology for Yate implementation of SS7
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|-
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* [[SS7 Implementation]]
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|[[MGCP_call_agent_module|MGCP]]
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List on SS7 components inplemented in Yate.
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|[[MGCP features]]
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* [[Interconnecting on SS7]]
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|-
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To connect to another SS7 network you will need to configure some parameters so the two sides speak the same language.
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|[[IAX|IAX2 (IAX1)]]
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* [[Signalling]]
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|[[IAX features]]
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This is a brief introduction on how the PSTN signalling modules are working together in Yate.
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|-
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|[[Yjinglechan|JINGLE]]
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|class="yate-content-right"|
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|[[JINGLE features]]
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* [[How_To's#SS7_Setups | Usage Scenarios]]
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|-
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Miscellaneous setups with various features: MTP2, MTP3, ISUP, CAMEL MAP, M3UA etc.
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|[[JABBER]]
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* [[M2PA VS M2UA]]
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|[[JABBER features]]
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A comparison between M2UA and M2PA
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* [[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.
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* [[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
 
|}
 
|}
  
  
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[[Mobile_networks|Mobile Networks]]
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 +
'''See also'''
 +
 
 +
*[[About VoIP]]
 +
*[[About PSTN]]
 +
*[[About ISDN]]
 +
*[[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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