Managing IP Centrex And Hosted PBX Service Quality

BY BOB MASSAD

IP Centrex (or hosted PBX) represents a major revenue opportunity for service providers over the next several years. Industry analysts expect the IP Centrex market to grow from approximately 13,000 lines in 2001 to approximately 10 million lines by 2008, partly from migration of the installed base of traditional Centrex users. IP Centrex is being deployed by traditional service providers, competitive carriers and long-distance service providers seeking to expand revenues by offering business services.

Key drivers for the IP Centrex market are the generally recognized benefits of Centrex services including lower upfront capital outlays and ease of use and maintenance versus a customer-owned, PBX-based system. Additionally, there are the benefits of IP telephony such as the lower costs of sharing a single, packet-based, broadband access system that can carry many simultaneous calls (and data packets) instead of dedicated copper wire pairs for each endpoint; the ability to build groups of Centrex users irrespective of their individual location who can benefit from features such as abbreviated dialing; and the large reduction in overhead to effect "adds, moves and changes."

An IP Centrex or hosted PBX system is a packet-based enterprise phone system in which the IP-PBX or call manager function is owned and operated by a service provider. IP phones are installed on enterprise user desktops, either provided by the enterprise or by the service provider. Typical IP Centrex systems employ SIP or H.323 to make calls and use "internal" supplementary Class services as if they had their own PBX. Currently, two basic types of IP Centrex solution are used:

1. GR303 gateways, which provide an IP front end to a Class 5 switch, thus making available the full range of (several hundred) Centrex features; and
2. Softswitches, which replace the CO's call switching function, and hence make it possible for new and competitive "pure IP" carriers to quickly introduce new services in competition with the RBOCs.

In each case, packetized voice traffic travels over both the enterprise LAN and the service provider's IP network. The enterprise LAN is owned and operated by the enterprise and guarded by sophisticated firewalls that protect the network from users on other networks. All packets traverse the firewall, which examines each and rejects those not meeting specified criteria. Consequently, the subscriber's LAN is made inaccessible to the service provider, which makes service-affecting problems occurring on the subscriber's LAN invisible.

Since the provider's and the subscribers' networks can be sources of quality degradation, both networks must be visible to the service provider to provide an "end-to-end" guaranteed or specified service level, which for this article is the call quality level. Although a service level agreement (SLA) may be in place to ensure the quality of the service at the demarcation point between the two networks, this does not guarantee the quality of the "inaccessible" enterprise network. There is also an implied responsibility for the quality of the IP Centrex voice that the service provider must meet, even though voice quality may be affected by problems on the enterprise LAN.

USING CALL QUALITY MONITORING AGENTS
Given our scenario, how can the IP Centrex service provider see through the enterprise firewall into the subscriber's network in order to maintain the quality of the service? This dilemma is a significant concern to IP Centrex providers and subscribers and has led to at least one major service provider delaying their IP Centrex deployment and another risking profitability by installing specialized hardware probes on the customer's LAN.

A method gaining significant support is the use of call quality monitoring agents that can be embedded directly into enterprise IP phones. These are capable of sending extended RTCP reports (RTCP XR) to provide call quality and diagnostic data in real time during the call. Then, the service provider collects this extended information and makes its own measurements at the demarcation point using agent technology. In this way, any problems related to the packet stream will be apparent and detectable in real time by the service provider, allowing them to be proactive in reporting and resolving problems. Further, comparison of call quality levels reported by the subscriber and measured within the provider's network supports quick problem isolation, i.e. "is the problem on my network or yours?"

RTCP is the "Real Time Control Protocol" implemented in conjunction with RTP, the "Real-time Transport Protocol" (see RFC 1889) used for voice, video and multimedia applications transmitted over an IP network. RTCP reports are routinely sent every 5-10 seconds by each IP phone during a call, traversing the same route as the RTP packets. This approach solves the firewall problem nicely, as firewall routers are already configured to allow both RTP and the associated RTCP packets to pass through.

The primary function of RTCP is to provide feedback on the quality of the session through the use of Sender and Receiver reports. However, the existing RTCP standard (RFC1889) has major shortcomings -- it does provide some performance metrics, however, these are grossly inadequate for monitoring the performance of VoIP services.

In order to address those shortcomings, Telchemy, Brix Networks, AT&T, UCSB, ShieldIP, University of Piere and Marie Curia, UT Dallas, Ericsson and others in the Audio/Video Transport Working Group in the IETF have developed a new approach that extends the capabilities of RTCP reports. Dubbed "draft-ietf-avt-rtcp-report-extns-06.txt" and available on the IETF Web site, RTCP XR supplements the statistics already contained in the report blocks with other and more detailed statistics that are tailored for the management of VoIP services. A complete list of contributors to the protocol is below.

Important RTCP XR metrics include:

1. Packet Loss Density
2. Packet Discard Density (due to jitter)
3. Burst Length (mS), Burst Density
4. Gap Length (mS), Gap Density
5. Round Trip Delay and End System Delay
6. R Factor
7. MOS - Listening and Conversational Quality
8. Packet Loss Concealment type
9. Jitter buffer type
10. Jitter buffer size (Min, Current, Max)

These statistics are critical in managing the IP Centrex service. The R Factor and MOS scores give a clear and simple grading of call quality. The service manager doesn't have to guess based upon isolated discrete statistics such as packet loss or jitter. The MOS or R Factor models aggregate and correlate the relevant information to provide an extremely accurate score for conversational and listener quality.

Other metrics are crucial to the troubleshooting process. For example, the most significant problem facing voice or video quality on an internal network is jitter. The discard density is a perfect indicator of jitter's impact while the jitter buffer metrics allow the potential misconfiguration of the jitter buffer to be detected. Packet loss burst metrics provide insight into another critical problem -- even if the overall packet loss rate for the call is low, there are often high density periods of loss that can be very annoying to the listener.

This powerful combination of software agents embedded into IP phones and real-time call quality/diagnostic data being transmitted between IP endpoints using RTCP XR provides a perfect solution for IP Centrex management. Service providers have the ability to monitor the quality of calls made by every IP phone, to measure quality at the desktop, to be immediately aware of problems affecting quality and to have the lowest possible cost for management infrastructure.

(Editor's note): A complete list of contributors to the RTP Control Protocol Extended Reports (RTCP XR) draft as taken from the draft:

Kevin Almeroth, UCSB
Ramon Caceres, ShieldIP
Alan Clark, Telchemy
Robert Cole, AT&T Labs
Nick Duffield, AT&T Labs-Research
Timur Friedman, Paris 6
Kaynam Hedayat, Brix Networks
Kamil Sarac, UT Dallas
Magnus Westerlund, Ericsson

Bob Massad is VP of marketing for Telchemy. Telchemy is a technology leader in the market for Voice over IP call quality monitoring and Quality of Service (QoS) management, and the "first" company to model the effects of time-varying-impairments. For more information, please visit their Web site at www.telchemy.com.