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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<!--{|class="wikitable"
 
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|colspan="12" style="text-align:center;font-weight:bold;" |Telephony modules in Yate
 
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|-
 
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|colspan="2" style="text-align:center;"|SIP
 
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|colspan="2" style="text-align:center;"|H323
 
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|colspan="2" style="text-align:center;"|MGCP
 
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|colspan="2" style="text-align:center;"|IAX2(IAX1)
 
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|colspan="2" style="text-align:center;"|JINGLE
 
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|colspan="2" style="text-align:center;"|JABBER
 
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|-
 
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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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|-
 
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|[[SIP in Yate]]
 
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|[[SIP Client]]
 
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|[[H323 channel module]]
 
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|[[H323 Gatekeeper And Multiple Endpoint Server]]
 
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|[[MGCP call agent module]]
 
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|[[Configuring Mediant MGCP]]
 
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|[[IAX|IAX channel]]
 
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|[[Yate as IAX server and YateClient as IAX client]]
 
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|[[Yjinglechan]]
 
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|[[Jabber Client With Jingle Yate Server]]
 
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|[[Jabber Client Module]]
 
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|[[Building Jabber server using Yate]]
 
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|-
 
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| [[SIP Listeners in Yate]]
 
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| [[SIP Features Module]]
 
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|
 
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|[[H323 To SIP Signalling Proxy]]
 
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|
 
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|[[Configuration for a Mediant MP-114 FXS analog gateway]]
 
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|
 
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|[[Connecting To IAXTEL]]
 
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|
 
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|[[Connecting to GMail]]
 
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|[[Jabber Server Module]]
 
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|[[Jbfeatures|Jbfeatures Module]]
 
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|-
 
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|[[SIP Messages]]
 
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|[[SIP Send DTMFs]]
 
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|
 
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|[[H323 Send DTMFs]]
 
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|colspan="8" rowspan="8"|
 
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|-
 
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|[[SIP Routing in Yate]]
 
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|[[SIP Attended Call Transfer In Cluster]]
 
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|
 
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|[[Compiling Yate with H323plus support]]
 
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|-
 
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|[[SIP Configuration File]]
 
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|[[SIP Flood Protection]]
 
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|colspan="10" rowspan="6"|
 
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|-
 
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|rowspan="5"|
 
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|[[SIP Generic Message]]
 
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|-
 
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|[[SIP SBC]]
 
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|-
 
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|[[xsip.generate]]
 
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|-
 
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|[[SIP query for CNAM and LNP]]
 
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|-
 
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|[[SIP NAT]]
 
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|-
 
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|}
 
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-->
 
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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="wikitable"
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{|class="yate-table"
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|class="yate-header-left"|'''About'''
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| style="width:500px; vertical-align:top; border:1px solid #fb9508;background-color:#fef0de;"
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|class="yate-header-right"|'''Features'''
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!About SIP
+
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|style="background-color:#fef0de;width:500px;border:1px solid #fb9508;vertical-align:top;"
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−
!SIP Features
+
 
|-
 
|-
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|colspan="2" class="article-content"|
+
|class="yate-content-left"|
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<div class="bullet">&nbsp;</div><div class="content"><font class="title">;[[SIP in Yate]]</font>  <br/>
+
* [[SIP in Yate]]
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;[[SIP in Yate]]
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SIP protocol in Yate
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: SIP protocol in Yate
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* [[SIP Security in Yate]]
−
; [[SIP Security in Yate]]
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TLS and SRTP in Yate
−
: TLS and SRTP in Yate
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* [[SIP Configuration File]]
−
;[[SIP Configuration File]]
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Main configuration file for SIP module in Yate.
−
:Main configuration file for SIP module in Yate.
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* [[SIP Methods]]
−
<!--;[[Listeners in Yate]]
+
How Yate processes SIP request methods and how to enable methods that are not handled by default.
−
:Listeners (TCP, UDP, TLS) specify on which network interfaces to listen. -->
+
* [[SIP Features Module]]
−
;[[SIP Methods]]
+
SIP features module that implements SUBSCRIBE and NOTIFY Methods.
−
:How Yate processes SIP request methods and how to enable methods that are not handled by default.
+
* [[SIP Routing in Yate]]
−
;[[SIP Features Module]]  
+
Route to a SIP channel and a SIP Line.
−
:SIP features module that implements SUBSCRIBE and NOTIFY Methods.
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* [[SIP Client]]
−
;[[SIP Routing in Yate]]  
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Implementation and SIP Client features in Yate
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:Route to a SIP channel and a SIP Line.
+
  
−
;[[SIP Client]]
+
|class="yate-content-right"|
−
: Implementation and SIP Client features in Yate
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* [[SIP Features Module]]
−
| style="width:500px; vertical-align:top; border:1px solid #aaa;" |
+
SIP features module that implements SUBSCRIBE and NOTIFY methods.
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;SIP Features
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* [[SIP Send DTMFs]]
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;[[SIP Features Module]]  
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How to do configurations related to DTMFs in SIP channel.
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:SIP features module that implements SUBSCRIBE and NOTIFY methods  
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* [[SIP Attended Call Transfer In Cluster]]
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;[[SIP Send DTMFs]]  
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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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:How to do configurations related to DTMFs in SIP channel.
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* [[SIP Flood Protection]]
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;[[SIP Attended Call Transfer In Cluster]]  
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Yate provides a protection mechanism against several types of SIP flood attacks.
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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 SBC]]
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;[[SIP Flood Protection]]  
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Describes how Yate can be used as a SIP session border controller.
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:Yate provides a protection mechanism against several types of SIP flood attacks.
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* [[xsip.generate]]
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;[[SIP SBC]]  
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Use this message to initiate the transmission of a SIP request.
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:Describes how Yate can be used as a SIP session border controller.
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* [[SIP query for CNAM and LNP]]
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;[[xsip.generate]]  
+
Feature used in a routing module that allows Yate to query CNAM and LNP databases over the SIP protocol.
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:Use this message to initiate the transmission of a SIP request.  
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* [[SIP NAT|SIP with NAT]]
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;[[SIP query for CNAM and LNP]]  
+
Resolving SIP traversal problem by Yate.
−
: Feature used in a routing module that allows Yate to query CNAM and LNP databases over the SIP protocol.
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* [[How_to_setup_chat_and_short_file_transfer_using_MESSAGE_Request_Method|How to use MESSAGE and PUBLISH requests]]
−
;[[SIP NAT|SIP with NAT]]  
+
Example on how to handle custom SIP requests.
−
<!--: Describes how Yate as a SIP Server can pass the voice (RTP) through a NAT.-->
+
−
:Resolving SIP traversal problem by Yate.
+
 
|}
 
|}
  
 +
== 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.
  
 +
{|class="yate-table"
 +
|class="yate-header-left"|'''About'''
 +
|class="yate-header-right"|'''Features'''
 +
|-
 +
|class="yate-content-left"|
 +
* [[H323 channel module]]
 +
Configuration file used to configure H323 protocol in Yate
 +
* [[OpenH323|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 - <font color="red">unstable version</font> to use with Yate
  
−
<!--
+
|class="yate-content-right"|
−
{| style="background:transparent;"
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* [[Yate as H323 GateKeeper and YateClient as H323 client]]
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| style="width:50%; vertical-align:top; border:1px solid #aaa;" |
+
Configure H323 Gatekeeper And Multiple Endpoint Server
−
;About SIP
+
* [[H323 To SIP Signalling Proxy]]
−
;[[SIP in Yate]]
+
Configure in Yate H323 To SIP Signalling Proxy
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:Describes the implementation of the SIP protocol in Yate and some SIP features.
+
* [[H323 Send DTMFs]]
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;[[Listeners in Yate]]  
+
Describes DTMFs send related to h323 channel.
−
:Listeners (TCP, UDP, TLS) specify on which network interfaces to listen.
+
−
;[[SIP Methods]]
+
−
:How Yate processes SIP request methods and how to enable methods that are not handled by default.
+
−
;[[SIP Features Module]]
+
−
:Describes SIP features module that implements SUBSCRIBE and NOTIFY methods
+
−
;[[SIP Routing in Yate]]
+
−
:Route to a SIP channel and a SIP Line.
+
−
;[[SIP Configuration File]]
+
−
:Main configuration file for SIP module in Yate.
+
−
;[[SIP Client]]
+
−
: Implementation and SIP Client features in Yate
+
−
 
+
−
| style="width:50%; vertical-align:top; border:1px solid #aaa;" |
+
−
;SIP Features
+
−
;[[SIP Features Module]]
+
−
:Describes SIP features module that implements SUBSCRIBE and NOTIFY methods
+
−
;[[SIP Send DTMFs]]  
+
−
: Describes 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 NAT|SIP with NAT]]
+
−
: Describes how Yate as a SIP Server can pass the voice (RTP) through a NAT.
+
−
|}-->
+
−
 
+
−
== H323 ==
+
−
 
+
−
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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{| style="background:transparent;"
+
−
| style="width:50%; vertical-align:top; border:1px solid #aaa;" |
+
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; About  H323
+
  
 
|}
 
|}
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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).
  
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{| style="background:transparent;"
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{|class="yate-table"
−
| style="width:50%; vertical-align:top; border:1px solid #aaa;" |
+
|class="yate-header-left"|'''About'''
−
; About MGCP
+
|class="yate-header-right"|'''Features'''
 +
|-
 +
|class="yate-content-left"|
 +
* [[MGCP call agent module]]
 +
This module implements the Call Agent part of MGCP allowing Yate to control analog or digital Media Gateways
 +
 
 +
|class="yate-content-right"|
 
|}
 
|}
  
Line 185: Line 88:
  
 
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.
 +
{|class="yate-table"
 +
|class="yate-header-left"|'''About'''
 +
|class="yate-header-right"|'''Features'''
 +
|-
 +
|class="yate-content-left"|
 +
* [[IAX]]
 +
Main configuration file for IAX channel module in Yate.
  
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{| style="background:transparent;"
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|class="yate-content-right"|
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| style="width:50%; vertical-align:top; border:1px solid #aaa;" |
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* [[How to configure Yate as IAX server|Yate as IAX server and YateClient as IAX client]]
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; About IAX
+
Setup to configure Yate as a IAX Server and YateClient as a IAX client
 
|}
 
|}
  
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== JINGLE ==
+
== JINGLE Protocol==
  
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<!--This module supports the Jingle protocol by implementing XMPP as an external component of another server.-->
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Jingle - Google talk/Google voice.
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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.-->
  
−
{| style="background:transparent;"
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{|class="yate-table"
−
| style="width:50%; vertical-align:top; border:1px solid #aaa;" |
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|class="yate-header-left"|'''About'''
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; About JINGLE
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|class="yate-header-right"|'''Features'''
 +
|-
 +
|class="yate-content-left"|
 +
* [[Yjinglechan| JINGLE Module]]
 +
How to configure JINGLE module in Yate
 +
 
 +
|class="yate-content-right"|
 +
* [[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
 
|}
 
|}
  
−
==JABBER or XMPP(Extensible Messaging and Presence Protocol) ==
+
==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).
 
Extensible Messaging and Presence Protocol(XMPP) is a communications protocol for message-oriented middleware based on XML(Extensible Markup Language).
−
{| style="background:transparent;"
+
 
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| style="width:50%; vertical-align:top; border:1px solid #aaa;" |
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{|class="yate-table"
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; About Jabber
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|class="yate-header-left"|'''About'''
 +
|class="yate-header-right"|'''Features'''
 +
|-
 +
|class="yate-content-left"|
 +
* [[Jabber Client Module]]
 +
Configuring Yate as a Jabber Client
 +
* [[Jabber Server Module]]
 +
Configuring Yate as a Jabber Server
 +
 
 +
|class="yate-content-right"|
 +
* [[Jbfeatures|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.
 
|}
 
|}
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<!--
 
  
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{|class="wikitable"
+
==SS7 Protocol suite==
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!Telephony modules
+
 
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! Features
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[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'''
 
|-
 
|-
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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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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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* [[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]]
−
|-
+
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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|[[Yjinglechan|JINGLE]]
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|class="yate-content-right"|
−
|[[JINGLE features]]
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* [[How_To's#SS7_Setups | Usage Scenarios]]
−
|-
+
Miscellaneous setups with various features: MTP2, MTP3, ISUP, CAMEL MAP, M3UA etc.
−
|[[JABBER]]
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* [[M2PA VS M2UA]]
−
|[[JABBER features]]
+
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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