IP Exchange - The Final Piece For VoIP

It’s no secret that the world’s telephone communications are on a fast path from circuit switched to packet switched networks. Large businesses have embraced enterprise VoIP as a way to improve productivity, reduce costs and future-proof their in-house telephone systems. Small and medium businesses are taking a close look at SIP trunking as a way to get high quality telephone and broadband service as a bundle. Consumers have taken matters into their own hands by moving to Cable broadband-telephone bundles or third party VoIP services. Yet, the PSTN persists.

What’s holding up this house of cards? Right now every VoIP system and service is an island unto itself. You’ll more often than not find that on-net calls for any particular VoIP service are free. On-net means on that particular service, unrelated to location. This is because the provider owns the network and doesn’t have to pay anyone else to handle the traffic. They are making money through the monthly fee they charge their customers for the VoIP service itself.

The same is true for large companies with their own enterprise VoIP systems. One of the big cost advantages of VoIP is that you can eliminate the separate telephone network and use your data network to transport voice, data and video. Another savings comes from keeping your voice traffic on your own network, not just within an office but also between locations. Avoid the public telephone network and you avoid the per minute charges to use it.

Therein lies the rub. The only common link between all telephones on Earth is the PSTN or Public Switched Telephone Network. Its legacy is based on switching analog phone circuits to make the phone to phone connections. Long haul circuits have long since gone digital, but a very special type of digital. The phone conversations are digitized according to an international standard, G.711, and packed into rigidly synchronized channels called DS0s. A DS0 is 64 Kbps and is the same whether being transported on a T1 or PRI trunk line right on up to a fiber optic OC-768 cable. Cellular phones have their own network standards, but are designed to interface with the PSTN to hand-off traffic.

The heart of the PSTN is both its technical specs for connecting analog and digital conversations and its switching systems called SS7. Nowadays the switching signals have their own separate paths between switching offices in a hierarchy consisting of local central offices and tandem offices that connect the local offices into one large universal network. If you want to connect with any phone on the PSTN, you need to send the signals according to the SS7 protocol.

VoIP comes as an outgrowth of the computers networking industry rather than the telephone industry. Its technology is based on IP packets and SIP switching rather than pulse code modulation channels and SS7. As you might expect, you can’t directly connect one system to the other. Analog and VoIP phone systems, including business PBX systems, connect or “terminate” their calls to the PSTN. The PSTN SS7 switches take care of getting the conversations connected from one proprietary network to another. Each time a call has to traverse the PSTN, there is a small but significant per minute access charge.

TW Telecom, a major competitive carrier, is looking to give the telephone system a nudge in the direction of eliminating the SS7 switching step. According to a recent report in Connected Planet, they are asking the FCC to classify voice calls over IP as a telecom service rather than the current status as an information service. That would get incumbent telephone companies, the heart of the old Bell system, to connect phone calls on an IP to IP basis. The benefits would include higher voice quality by eliminating the protocol conversions and lower network compensation costs.

TW’s actions are the first step in a process of creating IP Exchanges, similar to Ethernet Exchanges that share traffic between network providers and extending the reach of all Ethernet networks. It’s a matter of standardizing protocols and fees so that you have a level playing field without technical hiccups. It’s not the end of the beginning, but the beginning of the end for SS7.

Signaling for Call Centers

Every telephone call has two components. The actual voice conversation and the signaling that controls the telephone network. Signaling might be considered just the mechanics that go on behind the scenes. But how you implement the signaling function can make a big difference in your call center performance.

On analog telephone lines, voice communications and signaling share the same pair of wires, called the local loop. These copper wires connect the telephone set all the way to the phone company central office. Audible tones called DTMF or touch-tones are used to dial phone numbers and respond to voice commands. Lifting the handset or pushing a talk button completes a circuit that tells the telephone company switching system that your phone is in use. Ringing is a different electrical signal from the telco office to the handset. Caller ID information is a digitized signal that tells you who is calling.

This type of signaling is called in-band or common channel signaling. One shortcoming of this methodology is that sharing the line usually means you can signal or talk, but not generally at the same time. What may be less obvious is that in-band signaling is a slow process. It's not so much how fast you can push those buttons to dial a call. It's how fast a calling system can dial numbers. Each tone has to persist long enough to be properly decoded at the other end. The process of connecting and disconnecting calls, called call setup / teardown, creates wait time that can add up over thousands and thousands of calls.

What about digital trunking? Substituting a T1 line for 24 analog phone circuits creates a more efficient connection path between your PBX system and your supplier's central office. It's also likely to save you money if you have more than 6 to 8 outside lines. But talk and signaling still share the same lines. On T1 lines it's called CAS or Channel Associated Signaling because the digital line is make up of individual channels. T1 CAS is another implementation of in-band signaling.

ISDN, Integrated Services Digital Network, changes all that. An ISDN PRI trunk, also called a T1 PRI, separates talk and signaling. All of the signaling is conveyed over one dedicated delta channel. The other 23 bearer channels carry one telephone conversation each. This methodology is called CCS or Clear Channel Signaling. It's also called out of band signaling. Advantages of ISDN signaling include faster call setup / teardown, marginally better voice quality since the talk channels can use all of their bits for digitized voice signals, and ability to support features such as ANI or or Automatic Number Identification or Caller-ID.

The telephone carriers went to clear channel signaling decades ago to improve the speed and efficiency of their networks. They use a protocol called SS7 for Signaling System Seven. SS7 connects all the local (Class 5) and long distance (Class 4 and above) switches through separate switching links. Very large call centers may find it advantageous to by SS7 switching services to connect to the public telephone network. Such services not only include call control, but may also provide access to national line information and calling name databases.

The new competitor to traditional switching for call centers is SIP or Session Initiation Protocol. SIP is used in VoIP telephone systems to handle call control. The VoIP equivalent to the ISDN trunk is the SIP trunk. SIP trunks are not channelized. Everything is transmitted in streams of packets. A SIP trunk can carry dozens of telephone conversations and their associated SIP signaling simultaneously. The service provider handles the conversion from SIP to SS7 or ISDN to terminate calls to the public telephone network.

What type of telephone trunking and signaling is best for your contact or call center? Let our telecom technology experts review your situation, make recommendations and get you the best pricing for your call volume.