Advanced Hardware Features - Revision history

Allyntree at 15:33, 13 June 2022

2022-06-13T15:33:36Z

← Older revision		Revision as of 11:33, 13 June 2022
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	== Advanced ==		{{DISPLAYTITLE:Advanced Hardware Features}}

			Advanced hardware features enables you to tune the system for particular applications or environments. To modify this, go to:


	~~Advanced hardware features allow to tune the system for particular applications or environments. To modify this, go to:~~
	<pre>Hardware -> TB00xxxx [Edit] -> Advanced Parameters</pre>		<pre>Hardware -> TB00xxxx [Edit] -> Advanced Parameters</pre>

	'''NOTE:''' The options below require a restart of the main telecom module. Once all the changes have been done, go to:		'''NOTE:''' The options below require a restart of the main telecom module. Once all the changes have been done, go to:
	<pre>Hardware -> Status -> TB00xxxx -> Reboot adapter [Yes] -> Apply States</pre>		<pre>Hardware -> Status -> TB00xxxx -> Reboot adapter [Yes] -> Apply States</pre>

Luc Morissette: Updated so that it looks better

2013-04-16T15:01:22Z

Updated so that it looks better

← Older revision		Revision as of 11:01, 16 April 2013
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	Advanced hardware features allow to tune the system for particular applications or environments. To modify this, go to:		Advanced hardware features allow to tune the system for particular applications or environments. To modify this, go to:
	<pre>Hardware -> TB00xxxx [Edit] -> Advanced</pre>		<pre>Hardware -> TB00xxxx [Edit] -> Advanced Parameters</pre>
	The options below require a restart of the main telecom module. Once all the changes have been done, go to:		'''NOTE:''' The options below require a restart of the main telecom module. Once all the changes have been done, go to:
	<pre>Hardware -> Status -> TB00xxxx -> Reboot adapter [Yes] -> Apply States</pre>		<pre>Hardware -> Status -> TB00xxxx -> Reboot adapter [Yes] -> Apply States</pre>
	<br>		<br>
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	'''Enable HTTP or telnet access''' are set to allow the usage of the direct HTTP or telnet access to the main telecom module. Most of the time this is not needed, so they can be disabled. When using the TMP platform, the user needs to ssh to the TMG unit before accessing HTTP or telnet.		'''Enable HTTP or telnet access''' are set to allow the usage of the direct HTTP or telnet access to the main telecom module. Most of the time this is not needed, so they can be disabled. When using the TMP platform, the user needs to ssh to the TMG unit before accessing HTTP or telnet.
			<pre>Hardware -> TB00xxxx [Edit] -> Advanced Parameters -> Enable HTTP access
			Hardware -> TB00xxxx [Edit] -> Advanced Parameters -> Enable Telnet access</pre>
	<br>		<br>

	=== V34 fax ===		=== V34 fax ===

	'''Enable V34 FAX''' feature allows V.34 fax feature, in addition to V.17 fax. V.34 fax allows for faster fax transmission speeds. This should be selected only if there are V.34 faxes, since it reduces slightly the amount of channels available for fax. Please check the TBVoipWizard.html tool in the tb-doc.zip package.		'''Enable V34 FAX''' feature allows V.34 fax feature, in addition to V.17 fax. V.34 fax allows for faster fax transmission speeds. This should be selected only if there are V.34 faxes, since it reduces slightly the amount of channels available for fax. Please check the TBVoipWizard.html tool in the tb-doc.zip package.
			To enable V.34 fax:
			<pre>Hardware -> TB00xxxx [Edit] -> Advanced Parameters -> Enable V34 FAX</pre>
	<br>		<br>

	=== CPU optimization ===		=== CPU optimization ===

	Depending on the application, it may be useful to optimize the CPU usage. Signaling stacks have the ability to run on either CPU of the main telecom module. It is possible to verify memory and cpu usage and optimize the load of these modules:<br>		Depending on the application, it may be useful to optimize the CPU usage. Signaling stacks have the ability to run on either CPU of the main telecom module. It is possible to verify memory and cpu usage and optimize the load of these modules. The default configuration for these modules for maximum performance is like this:<br>

	*'''CAS R2'''
	*'''ISDN'''
	*'''ISUP'''
	*'''M2PA'''
	*'''M2UA'''
	*'''M3UA'''
	*'''SIP'''
	*'''IUA'''
	*'''LAPD'''

			<pre>Hardware -> TB00xxxx [Edit] -> Advanced Parameters -> CAS R2 -> 60%
			Hardware -> TB00xxxx [Edit] -> Advanced Parameters -> ISDN -> 60%
			Hardware -> TB00xxxx [Edit] -> Advanced Parameters -> Load ISUP on secondary CPU -> Yes
			Hardware -> TB00xxxx [Edit] -> Advanced Parameters -> Load M2PA on secondary CPU -> No
			Hardware -> TB00xxxx [Edit] -> Advanced Parameters -> Load M2UA on secondary CPU -> No
			Hardware -> TB00xxxx [Edit] -> Advanced Parameters -> Load M3UA on secondary CPU -> No
			Hardware -> TB00xxxx [Edit] -> Advanced Parameters -> Load SIP on secondary CPU -> Yes
			Hardware -> TB00xxxx [Edit] -> Advanced Parameters -> Load IUA on secondary CPU -> No
			Hardware -> TB00xxxx [Edit] -> Advanced Parameters -> Load LAPD on secondary CPU -> Yes</pre>
	<br>		<br>

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	*'''Increase SIP Stack Allowed RAM''': this allocates more memory to the SIP module. This should always be selected on unit sold since 2008.		*'''Increase SIP Stack Allowed RAM''': this allocates more memory to the SIP module. This should always be selected on unit sold since 2008.
	*'''SIP Dynamic/Event/Signal memory size and SIP Signal Pool size''': This is to control individual memory allocation on different modules of the SIP stack.		*'''SIP Dynamic/Event/Signal memory size and SIP Signal Pool size''': This is to control individual memory allocation on different modules of the SIP stack. A value of 0 lets the system select the memory allocation.
	*'''Reduced timer''': Default timer for keeping call information from terminated SIP calls is 32 seconds. When the Start threshold is reached the timer will be reduced so that the memory is freed faster, until the Stop threshold is reached. ~~Default is [4], or 40%, for Start threshold and [5], or 50%, Stop threshold~~		*'''Reduced timer''': Default timer for keeping call information from terminated SIP calls is 32 seconds. When the Start threshold is reached the timer will be reduced so that the memory is freed faster, until the Stop threshold is reached.
	*'''Congestion''': When the Start threshold is reached, the SIP stack will start refusing calls with congestion (503) error code. ~~Default is [2], or 20%, for Start threshold and [4], or 40%, Stop threshold~~		*'''Congestion''': When the Start threshold is reached, the SIP stack will start refusing calls with congestion (503) error code.
	*'''Critical Congestion''': When the Start threshold is reached, the SIP stack will start dropping incoming calls silently. Default is [1], ~~or 10%~~, ~~for Start~~ threshold ~~and~~ [3], ~~or 30%,~~ Stop threshold		*'''Critical Congestion''': When the Start threshold is reached, the SIP stack will start dropping incoming calls silently.
			Default Configuration:
			<pre>Hardware -> TB00xxxx [Edit] -> Advanced Parameters -> Increase SIP Stack Allowed RAM -> Yes
			Hardware -> TB00xxxx [Edit] -> Advanced Parameters -> SIP Dynamic/Event/Signal memory size and SIP Signal Pool size -> 0
			Hardware -> TB00xxxx [Edit] -> Advanced Parameters -> Reduced timer -> Start threshold 4, Stop threshold 5
			Hardware -> TB00xxxx [Edit] -> Advanced Parameters -> Congestion -> Start threshold 2, Stop threshold 4
			Hardware -> TB00xxxx [Edit] -> Advanced Parameters -> Critical Congestion -> Start threshold 1, Stop threshold 3</pre>
	<br>		<br>

	=== TDM Companding <br> ===		=== TDM Companding <br> ===

	This selects the companding mode of operation of the VoIP and IVR DSP devices. ~~ulaw~~ to ~~Alaw~~ conversion will be automatically done if needed. Should be set to ulaw for countries like US and Canada.		This selects the TDM companding mode of operation of the VoIP and IVR DSP devices. uLaw to alaw conversion will be automatically done if needed.
			Should be set to ulaw for countries like US and Canada:
	<br>		<pre>Hardware -> TB00xxxx [Edit] -> Advanced Parameters -> TDM Companding -> uLaw</pre>
			Other countries should use:
			<pre>Hardware -> TB00xxxx [Edit] -> Advanced Parameters -> TDM Companding -> aLaw</pre><br>

	=== VOIP Codec Mode<br> ===		=== VOIP Codec Mode<br> ===

	VOIP codec '''Mixed mode''' (default) allows any vocoder to be used in any call. Changing to '''Fixed Mode '''allows more channels for some vocoders, but the count of each type of vocoder must be configured at startup. Please check the TBVoipWizard.html tool in the tb-doc.zip package to know how many channels you can get.<br>		VOIP codec '''Mixed mode''' (default) allows any vocoder to be used in any call. Changing to '''Fixed Mode '''allows more channels for some vocoders, but the count of each type of vocoder must be configured at startup. Please check the TBVoipWizard.html tool in the tb-doc.zip package to know how many channels you can get.<br>
			Default configuration:
			<pre>Hardware -> TB00xxxx [Edit] -> Advanced Parameters -> VOIP Codec Mode -> Mixed Mode</pre>

	== <br><br> Call rate limiting feature ==		== <br><br> Call rate limiting feature ==

Lucas Trottier: Uniformize CAS R2 naming

2011-12-13T15:53:30Z

Uniformize CAS R2 naming

← Older revision		Revision as of 11:53, 13 December 2011
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	Depending on the application, it may be useful to optimize the CPU usage. Signaling stacks have the ability to run on either CPU of the main telecom module. It is possible to verify memory and cpu usage and optimize the load of these modules:<br>		Depending on the application, it may be useful to optimize the CPU usage. Signaling stacks have the ability to run on either CPU of the main telecom module. It is possible to verify memory and cpu usage and optimize the load of these modules:<br>

	*'''~~CASR2~~'''		*'''CAS R2'''
	*'''ISDN'''		*'''ISDN'''
	*'''ISUP'''		*'''ISUP'''
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	*'''LAPD'''		*'''LAPD'''

	<br>		<br>

	=== SIP stack optimization ===		=== SIP stack optimization ===

Luc Morissette: Added Advanced Hardware and Call rate limiting features

2011-11-15T16:09:23Z

Added Advanced Hardware and Call rate limiting features

← Older revision		Revision as of 12:09, 15 November 2011
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	== ~~<br> Call rate limiting feature~~ ==		== Advanced ==

	The ~~goal~~ of ~~this feature is~~ to ~~continue processing ongoing calls normally in~~ the ~~event~~ of ~~call bursts from~~ the ~~network~~. ~~Overloaded calls will~~ be ~~dropped with congestion~~. ~~This will prevent overloading~~ the ~~system~~, ~~continue normal call processing and keep~~ the ~~system stable~~. ~~These call bursts are more likely from a SIP network~~, ~~where calls can come from many different devices~~. ~~Calls per second in the system~~ are ~~calculated every second~~. ~~If the call rate specified in the configuration is lower than this value~~, the ~~temporary call buffer will be used. To allow~~ for ~~temporary call bursts, temporary call buffers are created ("call buckets")~~. ~~If this temporary call buffer is emptied,~~ the ~~calls will be instantly dropped with error code congestion (SS7/ISDN: 42, SIP: 503, CAS: back to Idle state)~~. ~~There are temporary call buffers for each signaling interface (SS7, ISDN, CAS and SIP), for overall calls~~ in the ~~system, and per Network Access Point (NAP)~~. ~~Once incoming calls are accepted (not rate limited)~~, the ~~system will always allow~~ the ~~outgoing call~~ to ~~be executed~~. ~~Dropped calls due~~ to ~~congestion are showed in~~ the ~~CDRs~~, ~~as well as traces in the logs (specify which)~~.		Advanced hardware features allow to tune the system for particular applications or environments. To modify this, go to:
			<pre>Hardware -> TB00xxxx [Edit] -> Advanced</pre>
			The options below require a restart of the main telecom module. Once all the changes have been done, go to:
			<pre>Hardware -> Status -> TB00xxxx -> Reboot adapter [Yes] -> Apply States</pre>
			<br>

			=== Limit access to advanced interfaces ===

			'''Enable HTTP or telnet access''' are set to allow the usage of the direct HTTP or telnet access to the main telecom module. Most of the time this is not needed, so they can be disabled. When using the TMP platform, the user needs to ssh to the TMG unit before accessing HTTP or telnet.

			<br>

			=== V34 fax ===

			'''Enable V34 FAX''' feature allows V.34 fax feature, in addition to V.17 fax. V.34 fax allows for faster fax transmission speeds. This should be selected only if there are V.34 faxes, since it reduces slightly the amount of channels available for fax. Please check the TBVoipWizard.html tool in the tb-doc.zip package.

			<br>

			=== CPU optimization ===

			Depending on the application, it may be useful to optimize the CPU usage. Signaling stacks have the ability to run on either CPU of the main telecom module. It is possible to verify memory and cpu usage and optimize the load of these modules:<br>

			*'''CASR2'''
			*'''ISDN'''
			*'''ISUP'''
			*'''M2PA'''
			*'''M2UA'''
			*'''M3UA'''
			*'''SIP'''
			*'''IUA'''
			*'''LAPD'''

			<br>

			=== SIP stack optimization ===

			To better control the SIP stack memory usage, new configuration parameters were added.

	~~Configuration~~:		*'''Increase SIP Stack Allowed RAM''': this allocates more memory to the SIP module. This should always be selected on unit sold since 2008.
			*'''SIP Dynamic/Event/Signal memory size and SIP Signal Pool size''': This is to control individual memory allocation on different modules of the SIP stack.
			*'''Reduced timer''': Default timer for keeping call information from terminated SIP calls is 32 seconds. When the Start threshold is reached the timer will be reduced so that the memory is freed faster, until the Stop threshold is reached. Default is [4], or 40%, for Start threshold and [5], or 50%, Stop threshold
			*'''Congestion''': When the Start threshold is reached, the SIP stack will start refusing calls with congestion (503) error code. Default is [2], or 20%, for Start threshold and [4], or 40%, Stop threshold
			*'''Critical Congestion''': When the Start threshold is reached, the SIP stack will start dropping incoming calls silently. Default is [1], or 10%, for Start threshold and [3], or 30%, Stop threshold

	~~1) Overall call rate limitations on SS7/ISDN/CAS/SIP stacks~~		<br>

	~~Hardware -> TB00xxxx [Edit] -> Call Rate Limiting~~		=== TDM Companding <br> ===

	~~- Maximum calls per second: If~~ the ~~call rate is higher than this value on this signaling interface (or overall),~~ the ~~call burst buffer~~ will be ~~used~~.		This selects the companding mode of operation of the VoIP and IVR DSP devices. ulaw to Alaw conversion will be automatically done if needed. Should be set to ulaw for countries like US and Canada.

	~~- Maximum call bursts: If this temporary call buffer is emptied, the calls will be instantly dropped with error code congestion (SS7/ISDN: 42, SIP: 503, CAS: back to Idle state).~~		<br>

			=== VOIP Codec Mode<br> ===

			VOIP codec '''Mixed mode''' (default) allows any vocoder to be used in any call. Changing to '''Fixed Mode '''allows more channels for some vocoders, but the count of each type of vocoder must be configured at startup. Please check the TBVoipWizard.html tool in the tb-doc.zip package to know how many channels you can get.<br>

	2) Call rate ~~limitation per NAP, SS7/ISDN/CAS/SIP~~		== <br><br> Call rate limiting feature ==

	NAP ~~-> NAP_xxxx [Edit] -> Calls Rate Limiting~~		The goal of this feature is to continue processing ongoing calls normally in the event of call bursts from the network. Overloaded calls will be dropped with congestion. This will prevent overloading the system, continue normal call processing and keep the system stable. These call bursts are more likely from a SIP network, where calls can come from many different devices. Calls per second in the system are calculated every second. If the call rate specified in the configuration is lower than this value, the temporary call buffer will be used. To allow for temporary call bursts, temporary call buffers are created ("call buckets"). If this temporary call buffer is emptied, the calls will be instantly dropped with error code congestion (SS7/ISDN: 42, SIP: 503, CAS: back to Idle state). There are temporary call buffers for each signaling interface (SS7, ISDN, CAS and SIP), for overall calls in the system, and per Network Access Point (NAP). Once incoming calls are accepted (not rate limited), the system will always allow the outgoing call to be executed. Dropped calls due to congestion are showed in the CDRs, as well as traces in the logs (specify which).

	~~- Maximum calls per second~~		Configuration:

	- Maximum ~~Call~~ bursts		1) Overall call rate limitations on SS7/ISDN/CAS/SIP stacks
			<pre>Hardware -> TB00xxxx [Edit] -> Call Rate Limiting
			</pre>
			*'''Maximum calls per second''': If the call rate is higher than this value on this signaling interface (or overall), the call burst buffer will be used.
			*'''Maximum call bursts''': If this temporary call buffer is emptied, the calls will be instantly dropped with error code congestion (SS7/ISDN: 42, SIP: 503, CAS: back to Idle state).<br>

	~~- Maximum simultaneous incoming calls: Limits the total number of incoming calls on this NAP to this value~~		<br>

	- Maximum simultaneous outgoing calls: Limits the total number of outgoing calls from this NAP to this value		2) Call rate limitation per NAP, SS7/ISDN/CAS/SIP<br>
			<pre>NAP -> NAP_xxxx [Edit] -> Calls Rate Limiting
			</pre>
			*'''Maximum calls per second '''
			*'''Maximum Call bursts'''
			*'''Maximum simultaneous incoming calls''': Limits the total number of incoming calls on this NAP to this value
			*'''Maximum simultaneous outgoing calls''': Limits the total number of outgoing calls from this NAP to this value
			*'''Maximum simultaneous total calls''': Limits the total number of calls on this NAP (should be set only for SIP NAPs)

	~~- Maximum simultaneous total calls: Limits the total number of calls on this NAP (should be set only for SIP NAPs)~~		<br>

Luc Morissette: formatting

2011-11-14T19:30:10Z

formatting

← Older revision		Revision as of 15:30, 14 November 2011
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	Call rate limiting feature:		== <br> Call rate limiting feature ==
	The goal of this feature is to continue processing ongoing calls normally in the event of call bursts from the network. Overloaded calls will be dropped with congestion. This will prevent overloading the system, continue normal call processing and keep the system stable. These call bursts are more likely from a SIP network, where calls can come from many different devices.
	~~Calls per second in the system are calculated every second. If the call rate specified in the configuration is lower than this value, the temporary call buffer will be used.~~
	To allow for temporary call bursts, temporary call buffers are created ("call buckets"). If this temporary call buffer is emptied, the calls will be instantly dropped with error code congestion (SS7/ISDN: 42, SIP: 503, CAS: back to Idle state). There are temporary call buffers for each signaling interface (SS7, ISDN, CAS and SIP), for overall calls in the system, and per Network Access Point (NAP).
	~~Once incoming calls are accepted (not rate limited), the system will always allow the outgoing call to be executed.~~
	~~Dropped calls due to congestion are showed in the CDRs, as well as traces in the logs (specify which).~~

	Configuration:		The goal of this feature is to continue processing ongoing calls normally in the event of call bursts from the network. Overloaded calls will be dropped with congestion. This will prevent overloading the system, continue normal call processing and keep the system stable. These call bursts are more likely from a SIP network, where calls can come from many different devices. Calls per second in the system are calculated every second. If the call rate specified in the configuration is lower than this value, the temporary call buffer will be used. To allow for temporary call bursts, temporary call buffers are created ("call buckets"). If this temporary call buffer is emptied, the calls will be instantly dropped with error code congestion (SS7/ISDN: 42, SIP: 503, CAS: back to Idle state). There are temporary call buffers for each signaling interface (SS7, ISDN, CAS and SIP), for overall calls in the system, and per Network Access Point (NAP). Once incoming calls are accepted (not rate limited), the system will always allow the outgoing call to be executed. Dropped calls due to congestion are showed in the CDRs, as well as traces in the logs (specify which).
	1) Overall call rate limitations on SS7/ISDN/CAS/SIP stacks
	Hardware -> TB00xxxx [Edit] -> Call Rate Limiting		Configuration:
	- Maximum calls per second: If the call rate is higher than this value on this signaling interface (or overall), the call burst buffer will be used.
	- Maximum call bursts: If this temporary call buffer is emptied, the calls will be instantly dropped with error code congestion (SS7/ISDN: 42, SIP: 503, CAS: back to Idle state).		1) Overall call rate limitations on SS7/ISDN/CAS/SIP stacks

			Hardware -> TB00xxxx [Edit] -> Call Rate Limiting

			- Maximum calls per second: If the call rate is higher than this value on this signaling interface (or overall), the call burst buffer will be used.

			- Maximum call bursts: If this temporary call buffer is emptied, the calls will be instantly dropped with error code congestion (SS7/ISDN: 42, SIP: 503, CAS: back to Idle state).



			2) Call rate limitation per NAP, SS7/ISDN/CAS/SIP

			NAP -> NAP_xxxx [Edit] -> Calls Rate Limiting

			- Maximum calls per second

	~~2) Call rate limitation per NAP, SS7/ISDN/CAS/SIP~~
	~~NAP -> NAP_xxxx [Edit] -> Calls Rate Limiting~~
	~~- Maximum calls per second~~
	- Maximum Call bursts		- Maximum Call bursts

	- Maximum simultaneous incoming calls: Limits the total number of incoming calls on this NAP to this value		- Maximum simultaneous incoming calls: Limits the total number of incoming calls on this NAP to this value
	- Maximum simultaneous outgoing calls: Limits the total number of outgoing calls from this NAP to this value
	- Maximum simultaneous total calls: Limits the total number of calls on this NAP (should be set only for SIP NAPs)		- Maximum simultaneous outgoing calls: Limits the total number of outgoing calls from this NAP to this value

			- Maximum simultaneous total calls: Limits the total number of calls on this NAP (should be set only for SIP NAPs)

Luc Morissette: Added Call rate limiting

2011-11-14T19:28:11Z

Added Call rate limiting

New page

Call rate limiting feature:
The goal of this feature is to continue processing ongoing calls normally in the event of call bursts from the network. Overloaded calls will be dropped with congestion. This will prevent overloading the system, continue normal call processing and keep the system stable. These call bursts are more likely from a SIP network, where calls can come from many different devices.
Calls per second in the system are calculated every second. If the call rate specified in the configuration is lower than this value, the temporary call buffer will be used.
To allow for temporary call bursts, temporary call buffers are created ("call buckets"). If this temporary call buffer is emptied, the calls will be instantly dropped with error code congestion (SS7/ISDN: 42, SIP: 503, CAS: back to Idle state). There are temporary call buffers for each signaling interface (SS7, ISDN, CAS and SIP), for overall calls in the system, and per Network Access Point (NAP).
Once incoming calls are accepted (not rate limited), the system will always allow the outgoing call to be executed.
Dropped calls due to congestion are showed in the CDRs, as well as traces in the logs (specify which).

Configuration:
1) Overall call rate limitations on SS7/ISDN/CAS/SIP stacks
Hardware -> TB00xxxx [Edit] -> Call Rate Limiting
- Maximum calls per second: If the call rate is higher than this value on this signaling interface (or overall), the call burst buffer will be used.
- Maximum call bursts: If this temporary call buffer is emptied, the calls will be instantly dropped with error code congestion (SS7/ISDN: 42, SIP: 503, CAS: back to Idle state).

2) Call rate limitation per NAP, SS7/ISDN/CAS/SIP
NAP -> NAP_xxxx [Edit] -> Calls Rate Limiting
- Maximum calls per second
- Maximum Call bursts
- Maximum simultaneous incoming calls: Limits the total number of incoming calls on this NAP to this value
- Maximum simultaneous outgoing calls: Limits the total number of outgoing calls from this NAP to this value
- Maximum simultaneous total calls: Limits the total number of calls on this NAP (should be set only for SIP NAPs)