Advanced Hardware Features - Revision history

Luc Morissette: Updated so that it looks better

2013-04-16T15:01:22Z

Updated so that it looks better

Lucas Trottier: Uniformize CAS R2 naming

2011-12-13T15:53:30Z

Uniformize CAS R2 naming

Luc Morissette: Added Advanced Hardware and Call rate limiting features

2011-11-15T16:09:23Z

Added Advanced Hardware and Call rate limiting features

Luc Morissette: formatting

2011-11-14T19:30:10Z

formatting

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)

@@ Line 2: / Line 2: @@
 Advanced hardware features allow to tune the system for particular applications or environments. To modify this, go to:
-<pre>Hardware -&gt; TB00xxxx [Edit] -&gt; Advanced</pre>
+<pre>Hardware -&gt; TB00xxxx [Edit] -&gt; 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 -&gt; Status -&gt; TB00xxxx -&gt; Reboot adapter [Yes] -&gt; Apply States</pre>
 <br>
@@ Line 10: / Line 10: @@
 '''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 -&gt; TB00xxxx [Edit] -&gt; Advanced Parameters -&gt; Enable HTTP access
 <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.
+To enable V.34 fax:
 <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>
+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'''
 <br>
@@ Line 40: / Line 41: @@
 *'''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:
 <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 -&gt; TB00xxxx [Edit] -&gt; Advanced Parameters -&gt; TDM Companding -&gt; uLaw</pre>
 === 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>
 == <br><br> Call rate limiting feature  ==

@@ Line 23: / Line 23: @@
 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'''
 *'''ISUP'''
@@ Line 33: / Line 33: @@
 *'''LAPD'''
 <br>
 === SIP stack optimization  ===

@@ Line 1: / Line 1: @@
-== <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:
-Configuration:
+=== Limit access to advanced interfaces  ===
-) Overall call rate limitations on SS7/ISDN/CAS/SIP stacks
+'''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.
-Hardware -&gt; TB00xxxx [Edit] -&gt; Call Rate Limiting
+<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.
+=== V34 fax  ===
-- 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).
+'''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.
-) Call rate limitation per NAP, SS7/ISDN/CAS/SIP
+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>
-NAP -&gt; NAP_xxxx [Edit] -&gt; Calls Rate Limiting
+*'''CASR2'''
-- Maximum calls per second
+<br>
-- Maximum Call bursts
+=== SIP stack optimization  ===
-- 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
+) Call rate limitation per NAP, SS7/ISDN/CAS/SIP<br>
-- Maximum simultaneous total calls: Limits the total number of calls on this NAP (should be set only for SIP NAPs)
+<br>