IP Telephony: April 2008 Archives
As part of a continuing series on Office Communications
Server 2007, this article will discuss the product's native telephony
capabilities.
As I've asserted many times, Unified Communications (UC) is an architecture
that may include many functional components, enablers and features. Presence,
Unified Messaging, Click-to-Dial, Simultaneous Ring, and mobility are just some
of the attributes that are often associated with UC. All of those play an
important role in UC but the core functional component of UC remains telephony.
In the context of this article, telephony refers to all of the features and
functionality associated with an enterprise-class business voice service.
Modern telephony services are now IP-based, either in the form of a
premise-based IP-PBX, a hosted IP Telephony service or various hybrid models.
When Microsoft announced the release of OCS in the fall of 2007 there were many
who claimed that it signaled the end of the PBX and that IP-PBX vendors such as
Cisco, Avaya, Nortel, NEC, Siemens, Mitel and others would be seriously
impacted by Microsoft's entry into the VoIP market. OCS does have VoIP
capabilities, but for most medium to large enterprise, the product's native
telephony features and functionally is inadequate. OCS can integrate with
IP-PBXs to provide the telephony requirements necessary for business. However,
the integration is not elegant (in most cases it must be done via media
gateways) and other than basic telephony features such as call origination and
transfer, the bulk of the enterprise telephony functionality will be provided
by the PBX; not OCS.
What follows is a partial list of telephony features and functionality required
(or expected) by most medium and large enterprise that are not available from a
standalone OCS solution:
E911
E911(as a telephony feature) refers to the ability of a telephony system to accurately identify the
location of the 911 caller and to accurately deliver that information to the
correct emergency dispatch location. That can be tricky in the world of IP
Telephony since IP Phones can easily be moved from one location to another.
However, there are proven and reliable ways to address the challenge. While
there are a variety of approaches, all major IP Telephony vendors have E911
functionality natively or have integrated 3rd party products into their
solutions.
In some cases, it is acceptable to have one or more analogue circuits to
provide E911 functionality. Also, an IP-PBX system installed in a small office
may not require E911 functionality if it is connected to the PSTN directly
since the telephone company would take care of the 911 location and routing
functionality.
However, in any large enterprise environment, E911 functionality is mandatory.
I've heard the argument that whether E911 functionality is required or not is
based on regulatory requirements for the location in question: nonsense! A
telephony architect or implementer has a moral responsibility to ensure that a
caller can place an emergency call when needed and that the call will be routed
to the correct location with the correct information.
SIP Trunking
IP Trunking, commonly referred to as SIP Trunking (since SIP has become the de facto
signaling protocol), is a relatively new method of connecting IP Telephony
systems to the PSTN. Traditionally, and still the most common approach, IP
Telephony systems are connected to the PSTN via Media Gateways. These devices
also known as IP Gateways, PSTN Gateways or VoIP Gateways, convert VoIP from
the IP Telephony system to ISDN or analogue circuits from the PSTN.
IP Trunking has many advantages, over the gateway approach including the fact
that it can improve the quality of voice calls by reducing, or eliminating,
conversions from one audio encapsulation method to another. Other advantages
are cost, eliminating points of failure, maintenance and other advantages which
I have listed in one of my previous articles on the subject.
Remote Survivability
Remote Survivability refers to the ability of a site, geographically separated
from centralized call control or PSTN connectivity or both, to remain
functional even if the site becomes isolated from the rest of the telephony
system. For example, if the network link between a branch office and the IP-PBX,
located at a company's head office is severed, the branch office may have a
requirement for telephony services to remain functional. The level of required
functionality will be dictated by the business requirements. IP-PBX vendors use a variety of methods
to provide remote survivability to their IP Telephony solutions which involves
distributing some, or all call processing functionality and PSTN connectivity.
Music on Hold
Most enterprises expect their telephony systems to play music, or some other
source of audio, to callers when placed on hold or when a call is transferred.
It is a basic functional component of any legacy or IP-PBX. In fact, most
enterprise systems support Music on Hold from a variety of sources and in some
cases, custom audio announcements whose content may be based on geographic
location or other criteria.
Hunt Groups
A common feature of a PBX is the ability to have incoming calls directed to a
queue where calls will be answered by agents based on a variety of criteria
including but not limited to first available agent, least busy agent, round
robin and others.
Attendant Console
While becoming less popular, many organizations still require that a live
person process all incoming calls for an organization or an enterprise
department. An Attendant Console will usually support many incoming calls and
will provide the attendant with visual and audio prompts and queues to aid in
the efficient processing of calls.
Admission Control
Admission Control allows a telephony system to refuse call attempts or to redirect
calls when it detects that insufficient network resources are available to
provide a quality voice path. I've heard Microsoft representatives make claims
that Call Admission Control is not required, but for the reasons discussed in
this previous article, I disagree.
Standards
One of the benefits IP Telephony systems is that many vendors support industry
standards such as SIP for signaling, and CODECS such as G.711 and G.729. While
not perfect, interoperability among vendors has been steadily improving for the
last several years. As a result it is now possible, for example to install an
IP-PBX from one vendor and select lower cost IP Phones from a different
vendor. Not so with Microsoft since they have chosen to implement their own
spin on SIP and have chosen their own proprietary CODEC called RTAudio. To date,
the only supported OCS phones are a model by Nortel/LG and a model by Polycom.
Neither phone supports LLDP-MED, a link discovery protocol that can play an important role with respect to VLAN, DHCP and Power over Ethernet configuration. The proprietary nature of the product also excludes the possibility of using 3rd party softphones such as the very popular X-Lite.
Geographic Diversity
A benefit for many large organizations deploying IP Telephony is that the
any-to-any nature of IP networks allows IP Telephony systems to be designed
with a high degree of redundancy by separating the call control functions
geographically. The business continuity benefits of such architecture can be
very compelling. OCS does not support geographic diversity.
Conclusion
I remember when Cisco first got into the telephony market approximately 10
years ago. Relative to the features and functionality available from TDM-based PBXs
(not to mention service quality and reliability), their product had minimal
functionality. As a result, it took many years of product development for their
IP Telephony product to move beyond the early adopters and into the mainstream
marketplace we see today.
Microsoft will, I am confident, continually improve their telephony
capabilities, but like Cisco, it won't happen overnight. Meanwhile, the IP-PBX
will remain alive and well for quite some time. And while Microsoft adds
telephony features to OCS, the PBX vendors will continue to evolve and improve
their own products; both from a telephony and an overall UC perspective.
The PBX is alive and well. And while it will continue to evolve, the PBX and
its leading vendors won't be disappearing any time soon.
Rick McCharles
Unified Communications Practice Principal
RIC Services,
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OCS is a Unified Communications (UC) product that integrates with Microsoft applications. It provides voice, presence, web/audio/video conferencing and integration with messaging systems. OCS can also integrate with PBXs, both TDM and IP-PBX. The introduction of OCS in the fall of 2007 created a lot of market hype and in my view, confusion. Microsoft marketing did a great job of capturing media attention and giving the impression that OCS was a revolutionary new communications technology and that enterprise communications, from that point forward, would be transformed forever. So effective was the marketing campaign, that some large organizations halted their VoIP migration planning in order to consider how OCS might alter their strategy.
The intent of this article and more that will follow, is to separate hype from reality so that you can make informed decisions about UC and how OCS should fit into your UC strategy. This first in this series of OCS articles discusses the topics of Call Admission Control (CAC) and Quality of Service (QoS).
Call Admission Control and Quality of Service – Short Tutorial
In the context of telephony, CAC refers to a telephony system's ability to decide whether a call request should be allowed or not. In the world of IP Telephony, VoIP is the common method of transporting voice. Analogue voice is converted to a digital from, encapsulated in IP packets, and then transported across a data network. A data network is made of a series of network links all of which have limits with respect to the amount of data they can carry at any give time. When a network link is oversubscribed, devices responsible for sending the data across the links will randomly drop data, which is what they are designed to do.
Quality of Service (QoS) mechanisms can be designed into networks to ensure that during periods of congestion, certain types of data will get priority over others. In IP Telephony implementations, voice and telephony-signaling data packets are marked with a priority label that informs routers that, in the event of congestion, that they should give priority to the telephony packets. However, if the amount of prioritized traffic exceeds a links capacity, then the router has no choice but to randomly drop even this high priority traffic. If the high priority traffic is telephone conversations then the quality of all calls will be affected, not just the call that exceeded the links capacity. This is where CAC steps in to avoid this situation.
CAC approaches vary but in all cases it consists of the system not allowing more calls than a resource can support. The most robust method will, in real-time, discover the available bandwidth available across the entire path of a proposed new call, then decide if the call should be allowed, and reserve the resource for the duration of the call. If insufficient resources are available the system can refuse the call or send the call across a backup path (like a PSTN gateway). More basic systems require a human to configure the system in advance with the maximum number of calls that should be allowed across a particular network path regardless of how much of the resource is available at the time of the call request.
Some of the arguments that I've heard, usually from vendors that don't support CAC, is that a network can be engineered in advance so that the over subscription problem never occurs (design for worst case). For a number of reasons, that approach is not practical in large enterprise environments. For one thing, bandwidth is not free. Second large IP networks are often designed so that automatic rerouting can occur during failures. In many instances the backup links will have less bandwidth than the primary links. So, a system designed to allow a certain number of calls based on a primary link's bandwidth could over subscribe a backup link.
OCS Support for QoS and CAC
While OCS does support marking voice payload packets for priority network treatment (DiffServ through DSCP) that functionality is not enabled by default. OCS has no Call Admission Control functionality. Microsoft claims that QoS and CAC are not required to ensure voice quality in an OCS environment. Microsoft uses a proprietary CODEC called RTAudio. The RTAudio codec is designed to detect network congestion or quality issues and to modify its bandwidth requirement accordingly. Also, the CODEC has an algorithm that allows it to send multiple duplicate packets to increase the probability that packets will arrive at their destination when network problems are encountered. As a result Microsoft argues that QoS and CAC are not required.
In my view, that logic is fine for Internet Telephony solutions. That is, if you’re transporting voice traffic across the Internet, then a CODEC that modifies its data usage and sends multiple copies of packets, makes sense. When transporting VoIP across the Internet, QoS markings will be ignored. In addition, there is no practical way of detecting in advance of call setup whether sufficient bandwidth is available along the entire call path. Even if one could determine that in advance, there is no way of ensuring that the desirable network conditions will persist for the duration of the call. And, if the fact that your throwing more packets at the problem, adversely affects other Internet traffic, who cares? One has little or no control of network conditions of random source and destination paths across the Internet. So it makes sense to use whatever means are at your disposal to ensure that your traffic gets through, and there is no point in employing mechanisms that won't make any difference.
But that logic does not fly in enterprise class telephony environments. First, while it is true that the RTAudio CODEC adjusts its bandwidth usage based on network conditions, it does not accomplish this instantaneously and therefore voice quality may be affected during the transition period since the algorithm starts with the assumption that there are no network issues. Secondly, while the CODEC can reduce its bandwidth usage, it does not reduce it to zero. When not using redundancy, the CODEC can reduce its consumption from 45Kb/s to 15Kb/s. Additionally, redundancy (sending multiple packet copies) may exacerbate a network congestion issue and potentially adversely affect competing enterprise traffic.
Additionally, what if WAN link is heavily congested? Should the system continue to process all requests across that link, even if the RTAudio CODEC is not able to compensate sufficiently to ensure adequate quality? With no CAC capability, this is exactly what OCS would do in that situation. Also, what if the business requirements dictate that a high fidelity CODEC is mandatory? The OCS approach would not be able to meet this requirement under congested network conditions. A properly engineered QoS solution with associated CAC however, could.
Conclusion
An IP Telephony solution, including OCS, when deployed in a typical large enterprise environment, cannot practically guarantee consistent voice quality without incorporating QoS and CAC mechanisms, end of story.
Notice, I did qualify the above statement with the word “practically”. It’s possible to engineer almost any technology deployment, no matter how deficient the technology may be, if one is free to ignore practical constraints such as business requirements, cost, manageability and scalability.
Rick McCharles
Unified Communications Practice Principal
RIC Services, Toronto, Ontario, Canada
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