Global Advantage Softswitch

The Global Advantage Softswitch is our premier softswitch platform. Offering our highest levels of control functionality in terms of architectural flexibility, signaling protocol breadth, control features and carrier-class manageability in a single platform. This product is our most scalable, full-featured softswitch solution that optimally delivers premium communication services across IP networks and is designed for large-scale, Tier 1 service provider deployments.

Below find information on key features in this solution. Modules can be enabled or disabled to meet customer specific requirements.

The Global Advantage comes with a media server in addition to all the features in the Advantage Softswitch.

System Design and Features:

Session Border Controller : SIP session border controller providing security for the internal network of the carrier. It also provides NAT traversal support.
- Integrated session border controller signaling & media control.
- Distributed media control only and/or signaling control.

VOIP to PSTN Gateway: Supports TDM and SS7 Networks connections.

H323 to SIP proxying: Giving you an added bonus on how you'd want to send your calls to the PSTN.

MEDIA SERVER FEATURES
- Music on hold
- Advanced IVR support and programming.
- Conference
- Call Parking
- Queue

Multi-protocol
- SIP
- H323(Through the media server)
- IAX(Through the media server)
- SS7(Through the media server)
- PRI (PRA) ISDN PRI (PRA) ISDN(Through the media server)

Codecs
- g711
- g723
- G.729A and G729B
- gsm
- g726

Architecture

• Modular Architecture: The modular architecture allows for plug-and-play module interface to extend the server's functionality.
• Scalability and High Density: Clusterable with each server able to handle over 1,000 to 5,000 call setups and 100,000 to 300,000 simultaneous calls depending on setup. The Softswitch can be used in geographic distributed-to-shares system load among several systems in order to drive optimum performance, reliability and scalability.
• Session Border Controller: SIP session border controller provides security for the internal network of the carrier. It also provides NAT traversal support.
  - Integrated session border controller signaling & media control.
  - Distributed media control only and/or signaling control.
• Media Gateway: Advanced media gateway handles the actual voice media. This system is responsible for converting normal E1/T1/PRI call signaling and media to SIP call signaling and media handled inside the Softswitch server.

 Softswitch Features

Radius support:
The Softswitch is compatible with all radius based billing systems. Radius support implementation for the AAA API from the core. 

Authentication:
The authentication mechanism offers protection against sniffing intrusion. Authentications can be handled using radius, database or configuration files

Dial Plan:
String translations based on matching and replacement rules. This module is highly configurable and also provides a programming interface for to do very complex dial plans.

Lease Cost Routing:
The least cost routing module like the Dialplan module is a programmable module and is very configurable. It offers dynamic routing so that changes to the routing take effect immediately. It is implemented as a programmable module that is depended on customer needs.
The basic LCR functionality does not need any customized programming and is built into the system.

Programmable LCR: Used for additional intelligent LCR functions like:
- LCR based call quality and price.
- Carrier Rates upload
- Region uploads
- Least Cost routing based on inter-state rates, intra-state rates and customized local calling areas.
- Conversion of rate plans based on LATA's, OCN's and Tiers into a routing table and rate plans based on NPA-NXX-X breakouts.

Basic LCR
When the rules are ordered based on costs. Dynamic Routing comes with many features regarding routing rule selection:
- prefix
- caller/group
- time
- priority

This module implements two capabilities:

• sequential forwarding of a request to one or more gateways
• sequential forwarding to contacts according to their q value

For the purpose of facilitating least cost routing of requests, each gateway belongs to a gateway group and each gateway group is associated with one or more <prefix, from pattern, priority> tuples. A gateway matches a request if user part of Request URI matches a prefix and caller's URI matches a pattern in a tuple that belongs to the group of the gateway.
Matching gateways are then ordered for forwarding purpose (1) according to longest user part match, (2) according to tuple's priority, and (3) randomly (prefix_mode = 0) or (1) according to gateway's priority and (2) randomly (prefix_mode = 1). In prefix_mode 0, prefix is a string of characters and in prefix_mode 1, prefix is a regular expression. From pattern is always a regular expression or empty. Empty from pattern matches anything. Smaller priority value means higher priority (highest priority value being 0).
When a gateway is selected, Request URI user part is stripped by the number of characters as specified by the gateways strip count. Subsequently, Request URI is rewritten based on gateway's URI scheme, tag, IP address, port, and transport protocol. Valid URI scheme values are NULL = sip, 1 = sip and 2 = sips. Tag is inserted in front of Request URI user part. Currently valid transport protocol values are NULL = none, 1 = udp, 2 = tcp, and 3 = tls.

PBX Dialing Support:
This allows uses of a particular group, company or department to have closed dialing within the group, locating users by their abbreviated code, besides their full identification. It offers functionality similar to Centrex. The relationship between users and their abbreviated codes, and their grouping is defined in a database table.

Call Control:
Allows one to limit the duration of calls and automatically end them when they exceed the imposed limit. Its main use case is to implement a prepaid system, but it can also be used to impose a global limit on all calls processed by the server.
Database Support:
The Softswitch has built in supports for Oracle, Postgres and MySQL and can support other databases using ODBC.

ENUM Support:
Enum module implements [i_]enum_query functions that make an enum query based on the user part of the current Request-URI. These functions assume that the user part consists of an international phone number of the form +decimal-digits, where the number of digits is at least 2 and at most 15. Out of this number enum_query forms a domain name, where the digits are in reverse order and separated by dots followed by domain suffix that by default is “e164.arpa.”. For example, if the user part is +35831234567, the domain name will be “7.6.5.4.3.2.1.3.8.5.3.e164.arpa.”. i_enum_query operates in a similar fashion. The only difference is that it adds a label (default "i") to branch off from the default, user-ENUM tree to an infrastructure ENUM tree.

INSTANT MESSAGE CONFERENCE:
This module offers support for instant message conference. It follows the architecture of IRC channels, you can send commands embedded in MESSAGE body, because there are no SIP UA clients, which have GUI for IM conferencing. You have to define an URI corresponding to im conferencing manager, where user can send commands to create a new conference room. Once the conference room is created, users can send commands directly to conference’s URI. To ease the integration in the configuration file, the interpreter of the IMC commands are embedded in the module, from configuration point of view, there is only one function, which has to be executed for both messages and commands.

PRESENCE AND MESSAGING SERVER:
This module implements jabber and connects users with heterogeneous IM networks. Via a Jabber server, it can connect to AIM, ICQ, MSN, and Yahoo! or XMPP networks. Each SIP user has to provide login details for each network he wants to interconnect with. It also is designed to provide a transparent gateway to XMPP, for IM and Presence, with scalability in mind.
The Softswitch also enables the exchange of instant messages between SIP clients and XMPP(jabber) clients.

LDAP Support:
This allows for using LDAP directory data in the SIP message routing script.
The following features are offered by the LDAP module:
- LDAP search function taking an LDAP URL as input
- Support for accessing multiple LDAP servers
- LDAP SIMPLE authentication
- LDAP server fail over and automatic reconnect
- Configurable LDAP connection and bind timeouts

OSP Support:
OSP is VoIP peering protocol and is the international standard for Inter-Domain pricing, authorization and usage exchange of IP communications.
The OSP module enables support of secure, multi-lateral peering using the OSP standard defined by ETSI (TS 101 321 V4.1.1). This module enables the server to:
- Send a peering authorization request to a peering server.
- Validate a digitally signed peering authorization token received in a SIP INVITE message.
- Report usage information to a peering server.
The major benefit of OSP for VoIP carriers is it provides a single, highly secure mechanism for managing diverse VoIP networks.  OSP provides a common interface between VoIP networks and the operations and billing support systems (OSS/BSS) used to manage VoIP networks.  By using a global VoIP to OSS/BSS interface, carriers are now enabled to build and manage multi-vendor, multi-protocol networks without impacting their central routing and billing operations.  The benefits are greater flexibility, vendor independence and lower operating costs.

PUA:
This module offers the functionality of a presence user agent client, sending Subscribe and Publish messages.

SIP-to-SMS IM gateway Support:
This module provides a way of communication between SIP network (via SIP MESSAGE) and GSM networks (via ShortMessageService). Communication is possible from SIP to SMS and vice versa. The module provides facilities like SMS confirmation--the gateway can confirm to the SIP user if his message really reached its destination as a SMS--or multi-part messages--if a SIP messages is too long it will be split and sent as multiple SMS.

Security Functions:
The basic architecture and design is built from the core  with security in mind. Every module is tested extensively to check for memory leaks, buffer and stack overloads know security holes.
There are also specific security modules that built into the system

Authentication
The authentication mechanism offers protection against sniffing intrusion. The module generates and verifies the nonces so that they can be used only once (in an auth response). This is done by having a lifetime value and an index associated with every nonce. Using only an expiration value is not good enough because as this value has to be of few tens of seconds, it is possible for someone to sniff on the network, get the credentials and then reuse them in another packet with which to register a different contact or make calls using the other’s account. The index ensures that this will never be possible since it is generated as unique through the lifetime of the nonce.

Peering
Peering module allows SIP providers (operators or organizations) to verify from a broker if source or destination of a SIP request is a trusted peer. In order to participate in the trust community provided by a broker, each SIP provider registers with the broker the domains (host parts of SIP URI's) that they serve. When a SIP server of a provider needs to send a SIP request to a non-local domain, it can find out from the broker using verify_destination() function if the non-local domain is served by a trusted peer. If so, the provider receives from the broker a hash of the SIP request and a timestamp that it includes in the request to the non-local domain. When a SIP server of the non-local domain receives the SIP request, it, in turn, can verify from the broker using verify_source() function if the request came from a trusted peer.
Verification functions communicate with the broker using an AAA protocol.
Comments and suggestions for improvements are welcome.

Permissions.
The module can be used to determine if a call has appropriate permission to be established. Permission rules are stored in plaintext configuration files similar to hosts.allow and hosts.deny files used by tcpd.
When allow_routing function is called it tries to find a rule that matches selected fields of the message.

DOS PREVENTION
The module provides a simple mechanism for DOS protection - DOS based on floods at network level. The module keeps trace of all (or selected ones) IP's of incoming SIP traffic (as source IP) and blocks the ones that exceeded some limit. Works simultaneous for IPv4 and IPv6 addresses.
The module does not implement any actions on blocking - it just simply reports that there is a high traffic from an IP; what to do, is the administrator decision (via scripting).

TLS
This module implements TLS related functions to use in the Switch.
Transport Layer Security (TLS) is a cryptographic protocol that provides security for communications over networks such as the Internet.
This allows Softswitch and Client communication without eavesdropping, tampering, and message forgery. TLS provides endpoint authentication and communications confidentiality over the Internet using cryptography. TLS provides RSA security with 1024 and 2048 bit strengths.

SNMP Support:
The Softswitch has an SNMP management interface  that provides general SNMP queryable scalar statistics, table representations of more complicated data such as user and contact information, and alarm monitoring capabilities.

SPEED DIAL
Per User server side speed dial.

SIP Session Timer support
The sst module provides a way to update the dialog expire timer based on the SIP INVITE/200 OK Session-Expires header value.
This allows for freeing of local resources of dead (expired) calls.

STUN SUPPORT.
A stun server working with the same port as SIP (5060) in order to gain accurate information.
STUN is an Internet standards-track suite of methods, including a network protocol, used in NAT traversal for applications of real-time voice, video, messaging, and other interactive IP communications.

Billing and Accounting Call Detail records (CDR’s): Real-time web access to Call Details Records and seamless integration with any radius or database based billing system.

Real-time rating engine

Tracing from CDR level to SIP protocol stack level

Multiple data-sources with consistent search and export capabilities

Trace calls at CDR level down to protocol level between data-sources

Login accounts can restrict access to CDR’s per user, domain or gateway

End-user web access to own Call Detail Records

Customizable end-user and administrative web front-ends

Link searches to trouble-tickets

Combined rating based on traffic, duration, application type and destination

Features Summary

• Music in hold
• IVR
• Conference
• Call parking
• Queues
• Plug and play module interface - ability to add new extensions, without touching the core, therefore assuring a great stability of core components
• Stateless and transactional statefull SIP message processing
• Support for UDP/TCP/TLS/SCTP transport layers
• IPv4 and IPv6
• Support for SRV and NAPTR DNS
• SRV DNS failover
• IP Blacklists
• Multi-homed (mhomed) and multi-domain support
• Flexible and powerful scripting language for routing logic
• Variables support in script - script variables, pseudo-variables (access to the SIP messages), AVPs (values persistent per SIP transactions)
• Management interface via FIFO file and Unix sockets
• Authentication, authorization and accounting (AAA) via database (Oracle, MySQL, Postgress, text files and other Databases through ODBC), RADIUS and DIAMETER
• Digest and IP authentication
• Presence Agent support (many additional integration features)
• XCAP support for Presence Agent
• CPL - Call Processing Language (RFC3880)
• SNMP - interface to Simple Network Management Protocol
• Management interface (for external integration) via FIFO file, XMLRPC or Datagram (UDP or unix sockets)
• NAT traversal support for SIP and RTP traffic
• ENUM support
• PERL Programming Interface - embed your extensions written in Perl
• Java SIP Servlet Application Interface - write Java SIP Servlets to extent your VoIP services and integrate with web services
• Load balancing with failover
• Least cost routing (LCR)
• Support for replication - REGISTER offer new functions for replicating client information (real source and received socket).
• Logging capabilities - can log custom messages including any header or pseudo-variable and parts of SIP message structure.
• Modular architecture - plug-and-play module interface to extend the server's functionality
• Gateway to SMS (AT based) – GSM gateway integration
• Multiple database backends - MySQL, PostgreSQL, Oracle, Berkeley, flat files and other database types which have unixodbc drivers
• Straightforward interconnection with PSTN gateways
• Dialog support (call monitoring, call termination from server side, call profiling)
• XMPP gateway-ing ( transparent server-to-server translation)
• Impressive extension repository - over 70 modules are included in Advantage Carrier Grade Softswitch repository

1.2 Scalability:

• Advantage Carrier Grade Softswitch can run on embedded systems, with limited resources - the performances can be up to hundreds of call setups per second
• The switch can be configured to handle calls in either two ways or both
  1) Statefull.The main use of statefull mode is so keep track of all call legs of a particular call in memory and to associate the calls into once call leg. This is costly in terms of memory and CPU. Some services inherently need state. For example, transaction-based accounting (module acc) needs to process transaction state as opposed to individual messages, and any kinds of ring group must be implemented statefully. Configured in this way the system can scale to handle over 2000 call setups per second, and 100 000 simultaneous calls.

2) Stateless:  In stateless mode, the server does not keep state and is beneficial in certain scenarios, for example outbound call centers, telemarketing companies, messaging etc where the call is usual one-way and does not get transferred around. Configured in this way the system can scale to handle over 5000 call setups per second, and 300 000 simultaneous calls.

• System can easily scale by adding more Advantage Carrier Grade Softswitch servers
• Advantage Carrier Grade Softswitch can be used in geographic distributed VoIP platforms
• Straightforward failover and redundancy

 

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