Toolpack monitoring sip simulator - Revision history

Abrassard at 18:40, 1 June 2017

2017-06-01T18:40:14Z

Lucas Trottier at 17:56, 17 May 2016

2016-05-17T17:56:47Z

Abrassard at 12:02, 5 May 2016

2016-05-05T12:02:03Z

Manon Bélanger: Fixed typos. Also specified that RTP generation is optional.

2016-04-22T13:02:18Z

Fixed typos. Also specified that RTP generation is optional.

Abrassard at 12:20, 22 April 2016

2016-04-22T12:20:54Z

Abrassard at 19:35, 21 April 2016

2016-04-21T19:35:39Z

Abrassard: /* Configuring SIP Simulators in Toolpack Web Portal */

2016-04-21T17:27:15Z

Configuring SIP Simulators in Toolpack Web Portal

Abrassard: /* Destination polling */

2016-04-21T17:26:53Z

Destination polling

Abrassard at 17:12, 21 April 2016

2016-04-21T17:12:47Z

Abrassard at 17:12, 21 April 2016

2016-04-21T17:12:11Z

@@ Line 1: / Line 1: @@
 [[Image:Monitoring_generator_overview.png|thumb|300px]]
-The Toolpack Monitoring "SIP Generator" is a behavior that can be attached to a call flow, and that will send simulated SIP messages to a destination host, simulating that a SIP call is being monitored.
+The Toolpack Monitoring "SIP Simulator" is a behavior that can be attached to a call flow, and that will send simulated SIP messages to a destination host, simulating that a SIP call is being monitored.
 The SIP simulator sends SIP messages, but does not expect any reply from the destination.

← Older revision		Revision as of 17:56, 17 May 2016
Line 115:		Line 115:

	For more information and step-by-step screen shots of the configuration of SIP Simulators, please refer to the [[Web Portal Tutorial Guide v2.9]].		For more information and step-by-step screen shots of the configuration of SIP Simulators, please refer to the [[Web Portal Tutorial Guide v2.9]].
		+
		+	[[Category:Tmonitor_OAM%26P\|SIP Simulator]]

← Older revision		Revision as of 12:02, 5 May 2016
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	Non-developers that use Toolpack system with built-in Gateway application can also use SIP Simulators and attach them to ISDN analyzers.		Non-developers that use Toolpack system with built-in Gateway application can also use SIP Simulators and attach them to ISDN analyzers.
−
−	~~In the Web Portal, configuring SIP Simulators is done by:~~
−	* [[Toolpack_monitoring_isdn_analyzer#Configuring_ISDN_analyzers_in_Toolpack_Web_Portal\|Creating Line Service Pairs, and ISDN analyzers]]
−	* Create IP interfaces (from the "IP Interfaces" section in the left menu)
−	** IP Interface(s) with RTP service (default "voip0" IP interface can be used here)
−	** Optional "Host" IP interface for service "H248/Radius", used as source IP interface for SIP messages generation
−	* Create a VOIP Media-only NAP (from the "NAPs" section in the left menu)
−	** Assign a "Port range" from the "RTP" IP interface created above
−	* Create SIP simulator
−	** Click on "Sip Simulators" section in the left menu
−	** Fill-out the required information in the page
−	** Create one or multiple SIP Simulator destinations (from the SIP Simulators configuration page)
−	* Associate this "SIP Simulator" to one or multiple [[Toolpack_monitoring_isdn_analyzer\|ISDN Analyzer(s)]] (from the ISDN Analysers' configuration page)

	For more information and step-by-step screen shots of the configuration of SIP Simulators, please refer to the [[Web Portal Tutorial Guide v2.9]].		For more information and step-by-step screen shots of the configuration of SIP Simulators, please refer to the [[Web Portal Tutorial Guide v2.9]].

@@ Line 1: / Line 1: @@
 [[Image:Monitoring_generator_overview.png|thumb|300px]]
-The Toolpack Monitoring "Sip Generator" is a behavior that can be attached to a call flow, and that will send simulated SIP messages to a destination host, simulating that a SIP calls is being monitored.
+The Toolpack Monitoring "SIP Generator" is a behavior that can be attached to a call flow, and that will send simulated SIP messages to a destination host, simulating that a SIP call is being monitored.
 The SIP simulator sends SIP messages, but does not expect any reply from the destination.
 It simulates both sides of the SIP call (simulating a sip "Invite" from one side, then "100-Trying" from the other side, etc.)
-It also sends a copy of the monitored legs audio using RTP packets, as indicated in the SDP of the simulated SIP message (Invite, 183-Progress and 200-OK).
+It can also send a copy of the monitored legs audio using RTP packets, as indicated in the SDP of the simulated SIP message (Invite, 183-Progress and 200-OK).
 == SIP simulator call flow ==
-Once attached to a call flow, the "sip simulator" behavior will identify the "main" legs of the attached call flow (one incoming and one outgoing), and watch for call state changes of these legs.
+Once attached to a call flow, the "SIP simulator" behavior will identify the "main" legs of the attached call flow (one incoming and one outgoing), and watch for call state changes of these legs.
 Corresponding SIP messages are generated.
@@ Line 24: / Line 24: @@
 === RTP generation options ===
-The Sip simulator will send RTP packets that contain a copy of the audio of the monitored call. These RTP packets are sent by a VOIP resource on the [[TMedia]] unit.
+The SIP simulator can send RTP packets that contain a copy of the audio of the monitored call. These RTP packets are sent by a VOIP resource on the [[TMedia]] unit.
 RTP generation options are:
-* The VOIP Media NAP to generate RTP from. This NAP defines:
+* The VOIP Media NAP to generate RTP from. "None" will disable RTP generation. This NAP defines:
 ** The source IP interface, and port range to use
 ** A default "profile"
@@ Line 103: / Line 103: @@
 To determine which destination is available, a "ping" will be made toward the destination SIP IP address.
-In addition the the "ping", a "TCP" polling can be performed, by attempting a TCP connection to the given IP address and port.
+In addition to the "ping", a "TCP" polling can be performed, by attempting a TCP connection to the given IP address and port.
 A destination is considered available if the "ping" and the "TCP connection polling" (if used) both succeed.
@@ Line 124: / Line 124: @@
 ** Click on "Sip Simulators" section in the left menu
 ** Fill-out the required information in the page
-** Create one or multiple SIP Simulator destinations (from the Sip Simulator configuration page)
+** Create one or multiple SIP Simulator destinations (from the SIP Simulators configuration page)
 * Associate this "SIP Simulator" to one or multiple [[Toolpack_monitoring_isdn_analyzer|ISDN Analyzer(s)]] (from the ISDN Analysers' configuration page)
 For more information and step-by-step screen shots of the configuration of SIP Simulators, please refer to the [[Web Portal Tutorial Guide v2.9]].

← Older revision		Revision as of 12:20, 22 April 2016
Line 1:		Line 1:
		+	[[Image:Monitoring_generator_overview.png\|thumb\|300px]]
	The Toolpack Monitoring "Sip Generator" is a behavior that can be attached to a call flow, and that will send simulated SIP messages to a destination host, simulating that a SIP calls is being monitored.		The Toolpack Monitoring "Sip Generator" is a behavior that can be attached to a call flow, and that will send simulated SIP messages to a destination host, simulating that a SIP calls is being monitored.

@@ Line 10: / Line 10: @@
 Corresponding SIP messages are generated.
-[[File:SIP_simulator_call_flow.png]]
+[[Image:SIP_simulator_call_flow.png|560px]]
 === Attaching to a monitoring call flow ===

@@ Line 115: / Line 115: @@
 In the Web Portal, configuring SIP Simulators is done by:
 * [[Toolpack_monitoring_isdn_analyzer#Configuring_ISDN_analyzers_in_Toolpack_Web_Portal|Creating Line Service Pairs, and ISDN analyzers]]
-* Optionally create IP interfaces (from the "IP Interfaces" section in the left menu)
+* Create IP interfaces (from the "IP Interfaces" section in the left menu)
 ** IP Interface(s) with RTP service (default "voip0" IP interface can be used here)
 ** Optional "Host" IP interface for service "H248/Radius", used as source IP interface for SIP messages generation

← Older revision		Revision as of 17:26, 21 April 2016
Line 107:		Line 107:

	The available destination with the highest priority will be chosen.		The available destination with the highest priority will be chosen.
		+
		+	== Configuring SIP Simulators in Toolpack Web Portal ==
		+	Developers can use SIP Simulators in their application by using class CTBCAFSipSimulator to define SIP simulator parameters and allocate SIP sender socket, and attach behavior CTBCAFCallBehaviorSipSimulator to their call flow.
		+
		+	Non-developers that use Toolpack system with built-in Gateway application can also use SIP Simulators and attach them to ISDN analyzers.
		+
		+	In the Web Portal, configuring SIP Simulators is done by:
		+	* [[Toolpack_monitoring_isdn_analyzer#Configuring_ISDN_analyzers_in_Toolpack_Web_Portal\|Creating Line Service Pairs, and ISDN analyzers]]
		+	* Optionally create IP interfaces (from the "IP Interfaces" section in the left menu)
		+	** IP Interface(s) with RTP service (default "voip0" IP interface can be used here)
		+	** Optional "Host" IP interface for service "H248/Radius", used as source IP interface for SIP messages generation
		+	* Create a VOIP Media-only NAP (from the "NAPs" section in the left menu)
		+	** Assign a "Port range" from the "RTP" IP interface created above
		+	* Create SIP simulator
		+	** Click on "Sip Simulators" section in the left menu
		+	** Fill-out the required information in the page
		+	** Create one or multiple SIP Simulator destinations (from the Sip Simulator configuration page)
		+	* Associate this "SIP Simulator" to one or multiple [[Toolpack_monitoring_isdn_analyzer\|ISDN Analyzer(s)]] (from the ISDN Analysers' configuration page)
		+
		+	For more information and step-by-step screen shots of the configuration of SIP Simulators, please refer to the [[Web Portal Tutorial Guide v2.9]].

@@ Line 47: / Line 47: @@
 * The simulated "from" IP address and port (can be "fictive" IP address representing one "side" of the call)
 * The simulated "to" IP address and port (can be "fictive" IP address representing one "side" of the call)
-* The SIP message text of each SIP message, built using [[toolpack_monitoring_sip_simulator_variables|well-known variables]]
+* The SIP message text of each SIP message, built using [[toolpack_monitoring_sip_simulator_variables|well-known variables]], and user-defined variables.
 ** From caller:
 *** SIP INVITE

@@ Line 63: / Line 63: @@
 *** SIP 200 OK (after BYE or CANCEL)
 *** SIP 487 Request Terminated
 For more information about available variables used to generate the SIP messages (SIP headers and body), read the following article: