Voice over Internet Protocol (VoIP) technology is currently revolutionizing telephony with the potential to dramatically reduce telecommunication costs while offering richer new functionalities. VoIP is slowly - but steadily - changing telephony in enterprise and residential markets globally.
However, the structural complexity of the VoIP environment (in contrast to that of the traditional PSTN) as well as diverse, still-maturing, signaling protocols, has made interoperability the primary challenge faced by vendors and purchasers. To understand how the interoperability challenge came about and what it may entail in the future, this paper will serve to address how VoIP telephony’s differs from traditional PSTN telephony.
Traditional telephony
Traditional, legacy PSTN systems transmit telephone conversations along a linear path over a generally homogeneous network known as a circuit-switched network. A fixed-line analog telephone is connected to the operator’s central office (CO) by a twisted-pair of copper wire. The CO identifies the signal - off-hook and dialed digits - which instructs it how to initiate, direct and terminate the connection, while routing the signal to the CO nearest to the receiving end. Before doing so, the CO converts the signal to a digital format so it can be sent effortlessly between telephony switches. The receiving CO then converts the signal back to analog and sends it to the telephone handset of the receiving party along the last mile. Once the telephone signal is relayed in this way the conversation proceeds smoothly and evenly along the same path allowing no danger of bottlenecks.
With legacy PSTN telephony:
- The entire network is optimized for voice:
- Continuity; subscribers are connected by a direct ‘line’ for the duration of the entire call
- Symmetrical bandwidth; subscribers transmit and receive over the same (unvarying) bandwidth
- Less vulnerability to degradation in voice quality
- Protocols used in traditional PSTN telephony have been developed for more than 100 years and are compatible with systems used by every operator worldwide.
VoIP telephony
The Internet, in contrast to PSTN, was designed for data communications. Sending voice over IP is different to sending it over the PSTN. On the Internet, voice is divided up and sent in data packets. Calls are individually routed between nodes over paths that are constantly re-routing in order to develop the fastest connection, in contrast to the single linear path of the PSTN. This random routing method can flaw call quality with delay, jitter and packet loss. VoIP is subject to the rules of data over IP (the 7-layer OSI model).
So why VoIP telephony?
Routing voice services over IP instead of via the PSTN offers a number of advantages to an enterprise or to a residential subscriber, primarily in the area of pricing. Instead of paying connection fees for each individual call, subscribers can pay their Internet Providers a flat rate for IP calls that travel along the same networks which are currently in use. This can translate into savings of thousands of dollars a month for distributed enterprises, and promises a quick ROI for enterprises making the transition.
In addition to the cost benefit, VoIP offers enterprises the potential to enrich their voice services with next-generation applications like VPNs and unified communications from different media. VoIP can also process a broader range of voice frequencies. Internet adaptability to voice services as well as advances in compression technology are enabling VoIP to reach a voice quality standard not inferior to (or better than) the PSTN.
Migrating to VoIP while preserving legacy PSTN equipment
Presently, most enterprises are still inextricably bound to PSTN equipment. Installing VoIP systems can require a prohibitively expensive investment in an IP PBX, IP telephones, IADs, IP switching equipment and media gateways. This solution may be suitable for a Greenfield (start-up) scenario but for an enterprise with an existing telecommunications infrastructure, it unfortunately means scrapping the installation base of legacy PSTN equipment. Therefore, in most cases, enterprises will opt for a hybrid solution, deriving all the benefits of VoIP without sacrificing their existing communication systems and without incurring the high cost of a full VoIP solution.
To achieve a hybrid solution, it is not enough for the equipment to be technologically superior, inexpensive or feature-rich. The equipment must be interoperability-rich. For a seamless hybridization, it is mandatory to use VoIP equipment that interoperates with the customer's existing legacy infrastructure.
AudioCodes' gateways, for example, enable enterprise customers to reliably port their existing telecommunications infrastructure to Next-Generation services while providing high levels of service on standards-certified and internationally homologated products, thus reducing deployment costs and ensuring a faster ROI.
VoIP equipment interoperating with internet components & services
Interoperability doesn’t end with the testing of VoIP equipment with enterprise customers’ existing legacy PSTN infrastructure. A more serious challenge than this exists. VoIP equipment needs to be interoperable with the dozens of different kinds of VoIP components and services developed – along with ongoing development - by VoIP OEMs and vendors of Softswitches, Call Agents, Call Managers, Proxies, Gatekeepers, IP PBXs, Soft Phones, VoIP Phones, VoIP Application Servers (such as Voice Mail, Unified Messaging, Media Servers/IVR), MCUs, MTAs, CMTSs, BTSs, MCSs, Firewalls and NATs . The way to synchronize and ensure reliable communications between new VoIP components and applications is to implement a standard protocol that should be followed by vendors worldwide. However, while this may sound easy, and probably will become so once the technology matures, VoIP is currently a battleground of competing vendors and standards bodies for control of the marketplace.
VoIP signaling protocols pose challenge to interoperability
Every VoIP call includes two parts: signaling protocol (which commands gateways to initiate, direct or terminate a call) and voice protocol. Voice interoperability is the easy part - RTP is presently accepted as the universal standard. Signaling is the principal challenge.
Four signaling protocols dominate VoIP telephony: H.323, SIP, MGCP and H.248/MEGACO.
H.323 and SIP are optimized for use with distributed network architecture. Based on ISDN and designed by the ITU-T (the most respected standards body in Europe), H.323 is the most mature standard. However, it is gradually being phased out in favor of SIP, which was developed by a group of computer enthusiasts and is now managed by the IETF, an open institute charged with developing and promoting Internet standards.
H.248/MEGACO and MGCP are primarily used in core networks by carriers and large network operators to control trunking gateways located between the PSTN and IP environments. H.248/MEGACO was developed according to IETF and ITU-T standards. MGCP was developed by the IETF.
Even if two gateways use the same signaling protocol (one of the four), it still doesn’t guarantee interoperability between them. Within a single protocol, there are multiple sub-standards, packages, versions and updates. Moreover, these always lag behind newly launched products and services in the market.
In addition, vendors often implement a protocol differently to one another, due to different interpretations of the specification. Furthermore, the standards contain hundreds of features, and vendors implement them according to different roadmaps, in a different sequence to what was specified in the standard and in a different sequence to one another. This generally doesn’t affect basic voice calls but it can affect supplementary services. To cap the above discrepancies, vendors sometimes choose to disobey the standards. Many, particularly the major players, are notorious for introducing their own unique protocols into their products in order to pressure customer enterprises loyal to them in the future.
Mixing & matching VoIP equipment with VoIP internet components & services
If vendors provided full end-to-end VoIP solutions, the deployment challenge would be easy to meet. Right now, very few vendors offer a comprehensive and interoperable solution. This means that an enterprise customer must mix and match components from different vendors in order to build its VoIP infrastructure, with the challenge of making them interoperate.
Achieving interoperability helps position VoIP products favorably in the market
Diversity of signaling protocols and services force manufacturers to ensure that their VoIP equipment interoperates with the most ubiquitous VoIP equipment in the market. Vendors periodically cooperate to perform interoperability tests on each other’s products and to work together in exchanging critical, non-proprietary information which can resolve interoperability issues quickly and inexpensively. At other times, manufacturers refuse to cooperate with each other, forcing one to acquire the other’s product and to run interoperability tests unilaterally. If a manufacturer cannot get another manufacturer’s cooperation agreement or to purchase their product for interoperability testing, the customer’s premises will be the arena in which they’ll meet. It is therefore highly advantageous for a vendor to employ a skilled interoperability team that can go out to the various arenas in the market, find incompatibilities and rectify them before deployment at customer sites.
How can interoperability activity be used to boost VoIP sales?
Vendors’ interoperability teams should distill the results obtained from every interoperability session into ‘Adaptation Packs’ (optimal parameter configurations for specific interworking) that can quickly and easily be implemented in future VoIP deployments. The larger the bank of ‘Adaptation Packs’ and the more operator-friendly it is, the greater the chances that Network Providers will prefer that vendor’s VoIP products over competitor products. An interoperability team’s power increases exponentially when R&D teams lend their resources for fixes, adaptations and improvements. By correctly integrating the efforts of the interoperability team with R&D, one’s VoIP products will quickly mature and achieve greater robustness and flexibility.
What’s needed to set up an interoperability lab?
Lab engineers must be thoroughly acquainted with the features and capabilities of the company’s products. They must also perfectly know how to operate and use them. It’s a significant advantage for lab engineers to be acquainted with the features and capabilities of other VoIP companies’ products. It is mandatory for the engineers to be acquainted with and updated on all Internet communication protocols. The lab must be equipped with a range of products, making it possible to simulate communications and services prevalent in the VoIP world. The lab must be connected to the Internet and all implications that would apply to an on-line environment. The more diversified a VoIP company’s products are, the more demands will be placed on the interoperability of the lab’s capabilities.
Conclusion
While VoIP has caught on quickly as a method to significantly reduce telecom expenses and enhance services, it is sometimes necessary to be reminded that the technology is still in its infancy stage. It is a fact that VoIP has been available for at least ten years, and quality has greatly improved over that time. However, compared to traditional telephony, which has matured over the space of more than 100 years, the market is still a long way from reaching maturity. Currently, the sector is reminiscent of the biblical tower of Babel, with each participant (and product) speaking a different language, and with only piecemeal solutions available to rectify their differences. Interoperability expertise helps to overcome those differences and will play a crucial role in the continued success of VoIP deployment for many years to come.