This article explores the Top-10 “cool” concepts of SIP (Session Initiation Protocol) Trunking which are:
1 - Integration of new hosted VoIP (Voice over Internet Protocol) solutions
2 - Integration of VoIP with IP-PBX (Internet Protocol - PBX-Private Branch Exchange)
3 - Migration to VoIP with older TDM (Time Division Multiplexed) PBX systems
4 - New IAD (Integrated Access Device) service provides dynamic bandwidth management
5 - IAD support for analog telephones
6 - IAD support for "hybrid" IP (Internet Protocol) and analog telephones
7 - Virtually unlimited incremental "slope" growth
8 - Integration with ISP (Internet Service Provider) business
9 - Advanced communication service provider services
10 - Integration with traditional and virtual call centers
“What is” SIP and SIP Trunking in 20 Words
Before jumping into the deep end of the pool, an introduction to SIP and SIP Trunking is important. Here’s the simplest, fastest definition we can come up with: SIP-Session Initiation Protocol is a "signaling" system for connecting, monitoring and disconnecting voice and multi-media communications sessions across the internet. If you prefer, Cisco explains SIP as, “defined by IETF (Internet Engineering Task Force) RFC (Request For Comment) 3261, SIP is a peer-to-peer, multimedia signaling protocol that integrates with other Internet services to deliver rich communications.”
For those technically-inclined, a SIP Trunk is a network interface device that recognizes SIP signals and can process these signals to other SIP devices. In other words, a SIP Trunk is digital data transmission link with SIP protocol signaling interfaces that recognizes SIP signaling and SIP media protocols using RTP (Real-Time Protocol) for media transmission (voice) and RTCP (Real-Time Control Protocol) for signaling and QoS (Quality of Service). 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.
1 - Integration of new hosted VoIP-Voice over Internet Protocol solutions
SIP is cool for companies of all sizes who are adding “Hosted VoIP” service and IP-Internet Protocol phones because it delivers VoIP without "fork lifting out" the existing PBX. Hosted VoIP service is much like having a hosted web site from an ISP (Internet Service Provider). Hosted VoIP offers many more features including a browser interface for feature changes, administration and support.
2 - Integration of VoIP with IP-PBX-Private Branch Exchange
SIP is cool for many customers with existing systems that are concerned about their investment in recently purchased, not yet-depreciated and other telephone equipment. That is, one of the inherent problems in expanding a PBX is the need to add station line cards, trunk cards and other equipment. This means that additional PBX equipment is added in a capital-intensive stair-step fashion. This often leads to under-utilized hardware or potential delays in waiting for new equipment to upgrade systems. SIP Trunking eliminates the need for onsite installation and can expand inexpensively, rapidly and remotely. Hosted VoIP coupled with SIP solutions offers customers the means to expand existing systems, implement branch, off-site and home offices as well as link directly to suppliers. SIP Trunking simplifies VoIP services if the customer has an IP-PBX which is a hardware system that converges (integrates) voice and data onto common all-IP-Internet Protocol network connections.
3 - Migration to VoIP with older TDM-Time Division Multiplexed PBX systems
SIP trunking is a cool solution to the painful but pleasant migration from old school TDM to new school VoIP. That is, before SIP trunking, nearly all enterprises had separate voice and data hardware and network connections. Voice PBX T-1 (1.544 MBPS-mega bits per second) trunks are connected to the traditional PSTN (Public Switched Telephone Network) for local and LD (Long Distance) calls and data T-1 circuits are connected to internet POP (Points Of Presence).
4 - New IAD-Integrated Access Device service provides dynamic bandwidth management
SIP Trunking is cool because it works with a traditional TDM PBX to simplify voice trunking with high speed data access. Coupled with an additional device called an IAD (Integrated Access Device), SIP Trunking dynamically manages data and voice packets with priority given to voice for high-quality performance. The IAD provides IAS-Integrated Access Service also known as CAS- Converged Access Services, a communications service that provides voice telephony connections as well as high speed data communications over the same T-1 circuit.
5 - IAD support for old analog telephones
On the other hand, if there is no PBX, the SIP Trunking IAD supports analog phones. Traditional telephone lines may still be needed for fax machines, credit card devices and other applications.
6 - IAD support for "hybrid" IP and analog telephones
SIP Trunking is cool because it supports a "hybrid" mix of IP and analog telephones which makes for an easy transition from “legacy” systems to next-generation IP services. This does not mean trading one telephone device for another with more features and buttons. It means the user can use a PC-based “softphone” – telephone features in software. While, “survey says,” that those more technically-inclined love softphones, many others in the organization also find the softphone equally useful. Anyone who prefers fewer rather than more devices is likely to find value in a softphone. Any “roady” may find a softphone to be the only solution rather than lug another piece of baggage around. Lastly, for the rest of us with fat fingers, it just saves a lot of time from misdialing because the number is on the screen.
7 - Virtually unlimited incremental "slope" growth
In addition, rather than traditional 24 channel "step growth," SIP Trunking supports virtually unlimited incremental or scalable growth sometimes referred to as N-way (uN-limited) growth – nothing is cooler than that. In other words, think of voice not as fixed channels but as data packets that share the bandwidth with other data applications. In this environment bandwidth is added only as required or when additional performance is required.
To expand on the concept of virtually unlimited growth, SIP Trunking allows a customer to oversubscribe their networking by using advanced compression techniques that increase capacity by 400% or more. There are a dozen different types of compression techniques such as G.711 and G.729 which have been approved by the international standards organization called the ITU-International Telecommunications Union (www.itu.org). Compression has been used in nearly all types of data networks to reduce bandwidth needs. For example, voice has been traditionally compressed (also referred to as sampled or quantized) at 64,000 BPS-Bits Per Second. However, in any conversation there are pauses intervals between words and even intervals within words that contribute little value to the conversation. In addition, most conversations are two-way which means only one person is talking at a time. Compression simply removes (compresses) unneeded or unused bandwidth making it available for other uses. Compression devices called CODECs-COmpression-DECompression (or Coder-DECoder) are highly advanced computer processing systems that can reduce voice to 8,000 BPS or 1/8 the bandwidth normally needed. Check with your provider for specific features and options. Without into greater depth on this subject, SIP Trunking with compression can bring considerable TCO-Total Cost of Ownership savings.
8 - Integration with ISP-Internet Service Provider business
Another business application is that SIP Trunking supports voice call processing from an ISP for hosted VoIP features and local-long distance call connections to the PSTN (Public Switched Telephone Network). This means via SIP Trunking, your ISP can provide high performance telephone service to its customers.
9 - Advanced communication services
New CSP’s-Communications Services Providers will bring to market new advanced multi-media and advanced SIP services. Here are some cool user applications enhanced by SIP Trunking:
- IM-Call Screening "presence" features are enhanced by SIP Trunking. Presence is a new concept that offers a wide range of features to let you connect, conference and be “present” when you can’t be there physically.
- "Event" notification can be enhanced with SIP Trunking for fire safety or business applications such as sporting, concert event or airline seat availability.
On demand “event” business meetings, training, broadcast announcements, call-to-meeting notifications, even reverse 911 (citizen notification services) are enhanced with SIP trunking.
- Integration of additional "third-party" developed SIP-enhanced services provides additional business and enterprise justification for SIP trunking. This means that organizations will be linked both vertically and horizontally via SIP trunking. Features and applications are just emerging, so stay tuned for exciting new features.
10 - Integration with traditional and virtual call centers
In addition, SIP Trunking supports applications such as automated (auto-dialing) outbound telemarketing or inbound order fulfillment. For the large enterprise with in-house or out-sourced call centers, SIP Trunking can connect them all together with free 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. In other words, whether you are a large or small business, with or without call centers, providing simple high performance communications is increasingly critical to success – very cool.
10.1 SIP Trunking and TCO
SIP trunking is more than a suite of cool concepts. It offers a profound set of business benefits and lowers overall TCO. Speaking of TCO-Total Cost of Ownership, SIP Trunking can offer significant lower TCO and operational cost-savings for enterprises by eliminating:
- The need for local PSTN gateways from costly separate voice ISDN BRIs (Basic Rate Interfaces) or PRIs (Primary Rate Interfaces) and data circuits
- Multiple voice and data hardware systems
- Separate network management tools
- Conferencing and webseminar bridge services
- Domestic and international long distance charges
- Security risks through voice encryption
- Duplicate trunks for disaster backup (or rather add additional redundancy via multiple SIP gateways)
- Need to terminate via PSTN with ENUM-Electronic NUMbers (ITU standard E.164) services
SIP Trunking also opens up a vast array of new call processing concepts under development.
SIP Security
While SIP brings advancement in VoIP call connections, SIP faces the same security attacks at other IP protocols such as HTTP and SMTP such as malformed message attacks, buffer overflow attacks, DOS-Denial-of Service attacks, eavesdropping (hijacking), RTP session hijacking, injection of unauthentic RTP packets into existing RTP flows and other known and yet to be created attacks. Special firewall and other SIP protection systems are recommended.
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.
Cool Conclusion
The key benefits of SIP Trunking to the small or large enterprise are both cool and profound. For the large enterprise, reducing CAPEX-capital expenditures on multiple network gateways located throughout the world is also significant. That is, using provider/carrier gateways reduces corporate capital investment and operational costs. For the small enterprise, interconnection of individual offices with other providers, channel partners or home office workers reduces local trunk charges. There are other savings on carrier "per-minute" connections to local or long distance networks (which may depend on PUC (Public Utility Commission approval).
Lastly, with SIP Trunking, the IP media (voice) stream coming from within the enterprise stays as an IP media stream and passes to anywhere within the organization or across the boundary to another one via IP. This reduces the need for hardware media gateways at the enterprise edge and carrier edge (often referred to as the PSTN) completely.
While SIP Trunking is an emerging concept for many users, it now offers a broad range of really “cool” features and considerable business benefits available today.
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