Channel Associated Signaling (CAS)
Channel Associated Signaling (CAS) is in-band signaling usually implemented as robbed-bit signaling. CAS is supported for both T1 and E1 (including DS3) on the IMG.
The IMG supports interworking between ISDN and SS7.
- For Interworking between ITU ISUP and ISDN, IMG follows Q.699 recommendation.
- For Interworking between ANSI ISUP and ISDN, IMG follows T1.609 recommendation
Provides cost effective TDM to SS7 ISUP switch.
- Fallback procedures are not supported in IMG, hence any interworking related to that is also not supported.
- Supplementary services are not supported in the IMG
SS7 ITU to SIP Interworking: Calling Party Category (CPC) (10.3.3 ER1)
The CPC is a parameter that characterizes the station used to originate a call and carries other important state that can describe the originating party.
This feature allows for the sending of the SS7 ITU Calling Party Category (CPC) in the From Header of SIP messages. Only SS7 to SIP is supported.
There are two options to place the CPC parameter in the From header field
- user part
- parameter of the From URI
The UPDATE method allows a UAC to update parameters of a session, such as the SDP and session timers.
The UPDATE method allows a greater control over a SIP session including, but not limited to, the following parameters:
- SDP (for example, to set the media on hold during early media)
- Session timers (for example, to adjust call duration in a prepaid application)
Allows you to update parameters during a session when conditions change.
The SIP UPDATE method is to be accepted by the IMG without user intervention, and therefore cannot be disabled. There is no configuration involved.
SIP Session Timer
Provides a "keep alive" method for SIP Calls which allows you to better manage your resources in an abnormal situation such as network outage.
The RFC 3261 does not define a keep alive mechanism for the sessions it establishes. The result is that an UA will not always be able to determine whether a session is still active. For instance, when a remote party fails to send a BYE message at the end of a session, or when the BYE message gets lost due to network problems, the UA will not know that session has ended, thus will not release resource allocated for the session. To resolve this problem, the IMG supports the keep alive mechanism for SIP sessions defined in RFC 4028 Session Timers in the Session Initiation Protocol (SIP).
You configure the Session Timer in the SIP Session Timer pane. SIP Session Timer is enabled by default.
SIP Busy Out
Set timers to determine if gateway is responsive or unresponsive.
RFC 3261 SIP: Session Initiation Protocol, Section 11.
The IMG can monitor the status of external SIP gateways by sending periodic SIP OPTIONS messages. If the gateway does not respond in a configured amount of time the IMG will mark the gateway as down and attempt to re-route the call to a different gateway.
Allows you to monitor the status of external SIP gateways.
The IMG accepts user agent subscription requests (SIP SUBSCRIBE method) and the ability to respond to those user agents with the appropriate DTMF digit events via the SIP NOTIFY method. Only DTMF-events are currently supported.
Detect DTMF tones (##) in the middle of a call.
You can develop user-specific applications that reside on your network entity and have the ability to subscribe for event services supported by the IMG. If the network entity wants the ability to detect an entered DTMF digit from the TDM-side of a call to the IP side of a call, the entity can subscribe to the IMG for these events and receive SIP NOTIFY events containing the digit event.
Improves network reliability and supports additional call flows.
There are two types of responses defined by SIP that are provisional and final. Final responses convey the result of the request processing and are sent reliably.
There are certain scenarios in which the provisional SIP responses must be delivered reliably. For example, in a SIP/PSTN inter-working scenario, a loss of 180 or 183 messages cannot be afforded. To solve this problem, the SIP PRACK method guarantees reliable and ordered delivery of provisional responses in SIP.
In addition, SIP PRACK provides further opportunities for SDP offer/answer exchange mode because of its 3-way handshake design.
SIP Carrier Identification Code
Send Carrier Id Code (CIC) between SIP and SS7.
Gives you the ability to send mixed traffic over a trunk group and improve call routing.
This feature enables the IMG to receive and transmit the Carrier Identification Code (CIC) parameter between the SIP network and SS7. The CIC parameter is a three- or four- digit code used in routing tables to identify the network that serves the remote user when a call is routed over many different networks. The CIC parameter is carried in SIP INVITE requests and maps to the SS7 Request.
ISUP-OLI (ANSI Only)
The ISUP OLI (also know as II digits) parameter includes information that is used for carriers to determine the origin of a call. This information gets lost over SIP networks if not inter-worked properly. This feature allows carrying ANSI ISUP OLI Parameter from traditional TDM network into SIP and vice versa. This information is passed in the From: header of the INVITE message.
Translates phone numbers to SIP addresses.
Bypass PSTN for lower phone charges and faster connections.
The IMG supports ENUM E2U+sip to resolve an ENUM telephone number into a SIP URI.
ENUM facilitates the interconnection of systems that rely on telephone numbers with those that use URIs to route transactions. E.164 is the ITU-T standard international numbering plan, under which all globally-reachable telephone numbers are organized.
SIP-Based Load Balancing
This feature allows you to distribute SIP traffic between IMGs configured as “SIP Servers” using virtual IP Addresses and a SIP load balancer.
Improved scalability and fault tolerance.
SIP CODEC Negotiation Priority
This feature allows you to configure whether the IMG or the remote gateway takes priority when selecting a codec.
The feature gives you the flexibility to choose CODEC priority on either IMG or the far end gateway.
You configure CODEC Negotiation Priority to either Local or Remote in the SIP Profile pane
SIP REGISTER “Expires” Timer Control
This feature gives you the ability to control the expiration of a registration request sent from the IMG to a Remote SIP User Agent. The default is 3600 sec.
You can change the default to a value between 10 and 7200 sec using the Registration Expiration Interval field in the External Gateway pane.
SIP 305 Redirect Response
The IMG now supports the 305 Use Proxy message.
Enables re-direct to Proxy Server.
SIP Call Hold
This feature allows the IMG to process a re-INVITE from a SIP endpoint that places a call on hold or releases a hold. This addition complements the current support for SIP Hold (via 0.0.0.0 ip address) by supporting RFC 3398 section 9, allowing for the proper Interworking of hold information between SIP and SS7.
ITX Message (SPIROU) Support (10.3.3 ER1)
SIP 3PCC (Third Party Call Control) (10.3.3 ER1)
The IMG supports the establishment of a session by a third party "controller" such as an application server or a Session Border Controller (SBC), allowing the IMG to integrate with services like auto attendant, conferencing, and unified messaging.
Fully Qualified Domain Name Support (10.3.3 ER2)
This feature allows you to manage IMG IP addresses via DNS for greater flexibility and security than hard-coded alternatives in SDP and SIP.
You configure outgoing FQDN options in the Outgoing Fully Qualified Domain Name field of the IMG 1010 - SIP Profile - 10.5.3 .
The radius dictionary file has been updated to support this feature. Customers using RADIUS should update the dictionary.cantata file they are using with the latest one located in the GCEMS install under ‘/opt/cantata/common/radius‘
On a freeRADIUS server this file should be copied into ‘/usr/share/freeradius’ directory.
The latest version of dictionary.cantata is v1.3:
# IMG release 10.3.3 ER2+
# Version. $id: dictionary.cantata,v1.3 2007/06/29 $
Support for multiple SIP 183 prior to 200 OK (10.3.3 ER2)
The IMG supports receiving multiple SIP 183 responses prior to a 200 OK for an INVITE request, when SIP responses can potentially be received from multiple remote user agents. A proxy or an Application server that receives an INVITE from the IMG can then fork the request to multiple destinations.
Multiple 183 responses are used in Follow-me services, and request forking scenarios. Applications that can benefit from the IMG handling multiple 18x responses are:
- Application Servers
- Proxy Servers
Re-INVITE if switching into Modem Bypass Mode (10.3.3 ER2)
With this feature the IMG will send a re-INVITE to the far-end when it detects modem traffic and switches the RTP into Modem Bypass mode over G.711. The IMG will use the bypass codec type specified in the associated bearer profile (u-law or a-law).
Incoming Re-INVITEs during modem calls are accepted automatically.
This feature is disabled by default. You enable it with the Outbound Modem Triggers Re-INVITE field in the IMG 1010 - SIP Profile - 10.5.3.
SIP INFO DTMF Digit Relay (10.3.3 ER2)
With this feature the IMG supports out-of-band tone passage of a single DTMF digit (0- 9, *, #, a, b, c, d) using the SIP INFO method.
Configuring the Feature:
You enable this feature by selecting INFO DTMF digit relay in the Method field of the IMG 1010 - SIP DTMF Support pane.
See IMG 1010 - SIP INFO Method - DTMF for more information.
Keep Alive Timer
Set Time to Live parameter in Registration Request (RRQ)
Greater user control when using H.323 gatekeepers
This feature gives you the ability to control the Time to Live value in the RRQ from the IMG when registering with a gatekeeper. This is an optional parameter in the RRQ. If no Time to Live is present in the RRQ, the gatekeeper returns a value in the RCF which the IMG has no control over.
By inserting the Time to Live value in the RRQ, the user has control over it as the gatekeeper normally returns the value requested by the endpoint in the RRQ. If the gatekeeper returns a different value, that will be the value used by the IMG in subsequent lightweight RRQs to maintain registration.
Pre/Post 95 Support for GRS/GRA
This feature provides the capability to support both pre and post 1995 GRS (Group Reset)/GRA (Group Reset ACK) messages on a per CIC group basis (ANSI Only).
Enables different CIC groups in an SS7 stack to connect to different types of DPCs.
The IMG supports JT-ISUP, the SS7 standard for Japan (JT-Q.763).
See IMG 1010 - ISUP Group - JTISUP for more information.
Cross Connect (10.3.3 ER3)
This feature allows you to pass SS7 Links through the IMG by establishing a cross connection between two TDM channels. These Channels can be on any spans in the same IMG that do not have an ISDN channel configured on them. These channels will not appear in any channel group.
Sending TNS (Transit Network Selection) Parameter for ANSI (10.3.3 ER3)
You enable the passing of the TNS in the IMG 1010 - SS7 Parameter Filter pane. If TNS is set to Pass and there is a Carrier ID (either in the incoming IAM or configured in the IMG 1010 - Translation Entry ), both Carrier Identification and TNS will be included in the outgoing message.
The Type of Network, Network ID Plan and Circuit Code sent in the TNS can be configured using the IMG 1010 - Advanced Carrier Number Translation pane. If a Carrier ID is entered in the IMG 1010 - Translation Entry but Advanced Carrier parameters are not configured, the Type of Network is set to National Network, Circuit Code is set to 0, and Network ID Plan depends on the number of Carrier ID digits.
Ability to assign Net5 Q.931 variant on T1 ISDN D-channels (10.3.3 ER2)
This feature allows you to configure Net5 Q.931 variant on a T1 ISDN D-Channel. This is required, for example, when connecting to carriers in South America where a far end using E1 is terminated in the US on T1s.
Enhanced Variant Support (10.3.3 ER2)
See the Supported Variants in the IMG 1010 - ISDN Features topic for a complete list of supported variants.
IMG 1010 - ISDN D Channel pane: Base Variant field
The G.729E/G codec is a low-bit rate codec that can support greater voice quality than standard G.729, using improved compression algorithms.
G.729E/G is available in VoIP Resource Profiles 6 and 7.
Better voice quality than G.729A.
G.723 Maximum Packet Size
The maximum packet size for the G.723 codec has been increased to 90.
Handle larger payloads for reduced bandwidth markets.
Modifiable RTP Base Port Number
This feature allows you to define the base RTP port value on a per module basis. The port for both VoIP modules can be set to any value in the range 8000-62462 in multiples of 2.
To modify the RTP port range in an existing configuration file, you must delete the VoIP module object under the Facility object and then re-create the object and change the port number.
This feature makes it easier for users with fire walls to insert the IMG into your network, because the RTP port ranges can be configured to fit your guidelines as to what port ranges are left open and what are closed.
G.729 AnnexB Selection
The media attribute "Annexb=no" can be sent by the IMG in the SIP SDP when enforcing the use of the G.729a payload type.
The annexb setting is available for G.729 and G.729E payloads.
Note that the media attribute "Annexb=yes" is not sent by the IMG in a SIP SDP, as this value is implied when unspecified in the SDP.
Modify Cause Code for Re-Routing Purposes
The IMG will automatically attempt to re-route calls in response to the following cause codes: 42, 41, 34. Use this field to select up to 4 additional Cause Codes for which the IMG will re-attempt a new call, per trunk group.
This feature gives you more flexibility as to how to use the IMG 1010 under particular network conditions and configurations.
Use the Re-attempt Cause Code field to select up to 4 cause code values.
New call tracing will be added to indicate that a re-attempt has occurred for the selected Cause Codes
Local Number Portability (LNP)
Allows the IMG to route ported numbers for SIP and SS7. Routing on LNP is supported on ANSI only. Mapping of LNP parameters is supported on both ANSI and ITU.
Properly handle ported numbers.
Routing on Originating IMG
This feature allows you to route based on the originating IMG as well as the Dialed Number.
You can route a call from the same dialed number to different channel groups, depending on the IMG it comes in on.
You enable this feature by selecting an IMG in the Match IMG as Well field in the IMG 1010 - Route Entry pane.
Importing Route Tables using NFS
More efficient method for loading large route tables.
Translation of Null String (10.3.3 ER 3)
This features allows you to translate a Null originating number only. This allows you, for example, to append a prefix to the originating number if the originating number is empty.
Enhanced Channel Group Statistics (10.3.3 ER 3)
The IMG now reports the following channel group information in the Status Panel of the Channel Groups pane:
- channels (TDM and IP) used per channel group
- channels in answered state
- number of channels in idle state
- number of channels out of service
- number of channels in wait/l3Clear state
- number of channels in wait/OutsiezeAck state
- number of channels in all other states
- total number of channels
Support for the following MIBs has been added to the IMG SNMP offering.
This feature will introduce support for the DS0-MIB as described in RFC 2494
Support for the alarm MIB offers three major benefits:
- The ability to inspect the alarm tables and find out what is currently wrong with the system.
- The ability to model alarms in a generic way.
- The ability to send out generic traps when alarms are raised.
- alarmActiveState - This trap is sent out when an alarm is raised. It contains the model number of the alarm as well as a resource ID which identifies specifically what the alarm refers to (for instance for a link down alarm you would indicate which link it is).
- alarmClearState - This gets sent out when an alarm is cleared. It contains the same information as the other trap.
IMG now supports the new Interfaces MIB specified in RFC 2863
Support for this MIB provides several benefits:
- A view of control, signalling, and data ports
- Descriptive information about these ports
- Measurements of the flow of traffic including number of packets in and out
- Recording of errors such as numbers of packets dropped.
Managers used to query the switch should update their list of compiled MIBs to include RFC 2863.
The IMG will reject any sets sent to it from a Network Management System and will not support sections of the RFC which call for sets.
Support for 3 SNMP Managers (10.3.3 ER2)
The IMG now supports up to 3 SNMP Managers.
RADIUS VSA Names
The word Cantata now appears at the beginning of all Cantata VSA names. See Cantata RADIUS VSAs.
RADIUS for Pre-Paid Applications
The IMG now accepts and acts upon data received in RADIUS Authentication Response messages that the Radius Server may send pertaining to prepaid application. This will allow the IMG1010 to be used in a prepaid application environment.
You enable this feature in the IMG 1010 - RADIUS Client pane.
Note: Radius Prepaid Support Mode will be disabled if Radius Debug Mode is enabled. The two modes cannot be enabled at the same time.
Routing using RADIUS Pre-Paid Application (10.3.3 ER1)
The IMG supports the routing of calls to a channel group indicated by a RADIUS server. If Pre-paid Support is enabled and the IMG receives VSA 45: Cantata-trunk-grp-out from the RADIUS server, the IMG will skip the mid-stream routing process and route the call to the channel group indicated by the RADIUS server.
There is no longer a Development Mode for the GCEMS. When you start GCEMS it is automatically started in Production Mode.
Setting Local Time
This feature allows you to set the local time zone on a per IMG basis. All IMG functions that use time will use the local time.
The local time is also used in the Radius attributes that use time:
- setup time
- connect time
- disconnect time
See IMG 1010 - Setting Local Time for more information.
You can retrieve WebHelp updates from the Cantata Support site that will replace the WebHelp version launched from ClientView.