Sip Trunk Provider | Services & Solutions | Secure & Reliable

SIP Trunk Overview

SIP-Session Initiation Protocol is a real-time communication protocol for VoIP.

To summarize, with SIP Trunking, the IP media stream coming from within the enterprise stays as an IP media stream passes to anywhere within the enterprise or across the boundary of the enterprise to another enterprise via IP. This reduces the need for hardware media gateways at the enterprise edge and carrier edge (often referred to as the PSTN) completely and provides significant savings to the enterprise.

SIP-Session Initiation Protocol is a “signaling” system for connecting, monitoring and disconnecting connections across the internet. A SIP Trunk is a network interface device that recognizes SIP signals and can process these signals to other SIP devices. SIP Trunking is provided by a Softswitch or SBC-Session Border Controller which provides, among other things, signal processing, protocol conversion, transcoding conversion, call routing, QoS-Quality of Service, AAA-Authorization, Authentication and Accounting functions as well as switching control interface to and from gateways.
Is a signaling protocol for internet conferencing, telephony, presence, events notification (emergency calling) and instant messaging. SIP has also been expanded to support video and instant-messaging applications. The function of signaling is to connect, monitor, alter and disconnect communications sessions. If you prefer, establish, change and terminate sessions. SIP does not address message content. SIP also uses a series of signaling commands to provide common responses.
A telephony signaling protocol that is used to establish a “communications session or connection” such as a telephone call, IM-Instant Message, conference call or other type of communications on an IP-Internet Protocol network. SIP is a request-response protocol that operates like a “communications browser” protocol such as HTTP-Hypertext Transfer Protocol. SIP is the communications equivalent to such internet protocols such as HTTP and SMTP-Simple Mail Transfer Protocol (SMTP).
Uses a text-based programming language designed to perform basic call-control tasks, such as session call set up and tear down as well as signaling for features such as call hold, caller ID, conferencing and call transferring.
Is an OSI Model Layer 5-Session protocol and is Layer 4-Transport protocol independent. SIP can be used with any Transport Layer-4 protocol such as:

UDP-User Datagram Protocol used in connection less communications such as streaming) and
TCP-Transmission Control Protocol used in connection-oriented verified transmission data transactions. SIP works on any other lower Layer 3 and below protocol such as IP-Internet Protocol, Ethernet, Frame Relay, and ATM-Asynchronous Transfer Mode.

The VoIP in SIP

VoIP (Voice over Internet Protocol) is the next evolutionary step in telephone service from its beginnings over a hundred years ago as Centrex or Central office Exchange (where telephone service is provided via switching connections in the CO-Central Office). PBX-Private Branch exchange services emerged as customers with large organizations wanted to control services/features and provide internal or private systems administration of multiple carriers at their corporate or branch offices. With the advent of VoIP-Voice over Internet Protocol also referred to as IPT-Internet Protocol Telephony, new features, capabilities and integration with office and computing applications emerged.

The following is an introduction to key VoIP options:

1 – IAS-Integrated Access Service The primary benefit of IAS-Integrated Access Service is the seamless integration of existing data networking equipment plus prioritized voice with dynamic data bandwidth allocation. VoIP, local or long distance connections are completed by the carrier or provider.
2 – HIPT-Hosted Internet Protocol Telephony The primary benefit HIPT-Hosted Internet Protocol Telephony is an outsourced hosted solution like a hosted website of IP voice telephony with dynamic data. This is often referred to as “everything is data” networking.
3 – MIPT-Managed Internet Protocol Telephony The primary benefit of MIPT-Managed Internet Protocol Telephony is a provider administered service of all CPE-Customer Premise Equipment and data devices on the IP voice network to ensure performance and QoS.

SIP & Integrated Access Devices

SIP Trunking works with traditional TDM-Time Division Multiplexed PBX-Private Branch Exchange telephone systems to merger voice trunking with high speed data access. The IAD (Integrated Access Device) dynamically (automatically) manages data and voice packets with priority given to voice for high-quality performance.

IADs provide the service known as integrated access also known as converged access (flex services) a communications service which provides voice telephony connections as well as high speed data communications over the same T-1 circuit. While integrated access service has been around for more than five years, new dynamic bandwidth and other capabilities are now available. While the difference between integrated and converged access is often more marketing than actual, integrated access usually refers to as fixed allocation of bandwidth. For example, the carrier would provide up to 50% of the bandwidth or 12 channels for telephone connections and the remainder or 768 kilobits per second for internet or wide area data networking. With converged access, as shown here, bandwidth is not fixed but allocated based on need. Voice is prioritized for high quality of service or QoS. One carrier provides up to 10 high quality and up to 40 low quality voice calls with the remainder of the bandwidth available for data. This means if no one is the phone, the entire T-1 bandwidth is available for data communications. In addition, VoIP-Voice over Internet Protocol and other features are available depending on specific carrier offerings.

If there is no PBX, the SIP Trunking IAD supports analog phones. SIP Trunking also supports a “hybrid” mix of IP deskset telephones, PC-based softphone IP and analog telephones.

SIP Security

While SIP brings advancement in VoIP call connections, SIP faces the same security attacks as other IP protocols such as HTTP and SMTP such as malformed message attacks, SPIT-SPam over Internet Telephony, buffer overflow attacks, DOS-Denial-of Service attacks, eavesdropping, hijacking, injection of malicious RTP packets into existing RTP flows and other known and yet to be created attacks. Special SIP firewall and other protection systems are recommended.

Advanced techniques such as SRTP (Secure Real-Time Protocol) is one method to provide an additional level of security. SRTP is a Transport Layer 4 protocol which intercepts RTP packets and then forwards an equivalent SRTP packet on the sending port, and intercepts SRTP packets and passes an equivalent RTP packet up the stack on the receiving port. A single MKI-Master Key Identifier provides digital keys for confidentiality and integrity protection for both for the SRTP stream and the corresponding SRTCP stream. In addition, salting bits (keys) (added like adding salt to food) can be added to the MKI-Master Key Identifier to protect against pre-computation and time-memory trade of cipher/hacker attacks. This creates “hash code” (like chopped corn-beef hash) unreadable code characters with a nonce (time stamp or other randomly generated code or word). The MKI “salting” guarantees security against off-line key-collision attacks on the key derivation that might otherwise reduce the effective key size. IETF RFC-3711 is recommended for further reading.

SIP & QoS-Quality of Service

RTCP (Real-Time Control Protocol) packets are used to provide QoS measurement reports and other information. The VoIP RTCP-XR-eXtended Reports MRB-Metrics Report Block provides measurements (metrics) for monitoring quality of VoIP calls and conversations. These measurements include packet loss and discard metrics, delay metrics, analog metrics, and voice quality metrics.

The MRB (Metrics Report Block) reports individually on packets lost (discarded) on the IP channel as opposed to packets that have been received and then lost by the receiving jitter buffer. MRB reports on the combined effect of losses and discards which can be used to determine corrective actions on voice QoS. Network analyzers or probes gather RTCP-XR packets and provide reports for management evaluation. This report provides details on Delay, Packet Lost, Packet Discard, Jitter, R-Factor, MOS-Mean Opinion Score and other factors.

SIP Trunking & Old PBX-Private Branch Exchange Systems

One of the advantages of hosted VoIP service is that they can be added without “forklifting out” the existing PBX telephone system. However, the key point is that SIP Trunking and Hosted VoIP or hosted PBX are compatible strategies for user implementation. That is, SIP Trunking can be used at larger locations with hosted VOIP used for virtual or mobile users for maximum benefit and lower TCO-Total Cost of Ownership.

One of the inherent problems to expand a PBX is the need to add station line cards, trunk cards and other equipment are often required. That is, additional PBX equipment is added in a capital-intensive stair-step fashion. This often leads to under-utilized hardware. SIP Trunking eliminates the need for onsite installation and can expand inexpensively, rapidly and remotely.

SIP Un-Limited Growth

In addition to PBX hardware savings, there are savings in regard to network (bandwidth) connections. Rather than the traditional 24 channel “stair step growth,” SIP Trunking supports virtually unlimited incremental or scalable growth sometimes referred to as N-way (un-limited) growth. In other words, think of voice not as fixed channels but as data packets which share the bandwidth with other data applications. Only as needed is bandwidth then added.

SIP Trunking also allows a customer to oversubscribe (increase capacity) the number of voice calls by utilizing advanced compression techniques such as G.711 and G.729 CODECs-COmpression-DECompression (or CODer-DECoder) devices increasing capacity by 400% or more. Check with your provider for specific features and options.

SIP & TCO-Total Cost of Ownership

SIP Trunking can also provide significant lower TCO (Total Cost of Ownership) and operational cost-savings for enterprises by eliminating:

1- The need for local PSTN gateways from costly separate voice ISDN BRIs (Basic Rate Interfaces) or PRIs (Primary Rate Interfaces) and data circuits
2 – Multiple voice and data hardware systems
3 – Separate network management tools
4 – Conferencing and webseminar bridge services
5 – Domestic and international long distance charges
6 – Security risks through voice encryption
7 – Duplicate trunks for disaster backup (or rather add additional redundancy via multiple SIP gateways)
8 – Need to terminate calls via PSTN by using E.164 ENUM (internet telephone numbers) services
9 – And other organizational management costs.

SIP Trunk Service Area’s:

Road Map

Development of new SIP trunking software applications may provide even more significant savings.

Lastly, SIP Trunking also opens up a vast array of new call processing concepts not yet developed.

SIP Applications & Future Outlook – Something for Everyone

IM-Call Screening “presence” features are enhanced by SIP Trunking.

“Event” notification can also be enhanced with SIP Trunking for fire-public safety or business applications such as sporting-concert events, restaurant-airline seat availability or stock price monitors. On demand business meetings, training, broadcast announcements, call-to-meeting notifications, even reverse E911 are enhanced with SIP trunking. Integration of additional “third-party” developed SIP-enhanced services provides additional business and enterprise justification for SIP trunking.

SIP Trunking supports next-generation communications service provider applications such as automated (auto-dialing) outbound voice auto-dialed telemarketing, “event broadcast” emails/vmail or inbound touchtone order fulfillment. SIP Trunking supports on-net toll-free calling and conference calling. Inbound or outbound call centers can be connected for normal or overflow call processing. SIP Trunking supports on-net toll-free calling and conference calling. Inbound or outbound call centers can be connected via VTL-Virtual Tie-Lines for normal or overflow call processing.

Coming soon is IMS or IP Multimedia Subsystems the recognized international standard for IP interoperability, roaming, bearer user control, charging and security. SIP is one of the key protocols in an IMS system.