Mixing Voice & Date(Part Two)
By Dr. Peter Welcher, Senior Consultant-Instructor

In part one of this series (see The Network Monitor Volume 4, Number 1) we focused on the history and technology of voice networking. We also looked at the Cisco MC3810 and Voice over Frame Relay on ATM (VoFR, VoATM).

In this part two of our voice coverage we will look at Voice Over IP and design options.

Chesapeake feels that this technology is becoming very important and Chesapeake became the first Cisco-Certified Training Partner to offer the Multiservice Concentrator Configuration & Monitoring (MCCM) course. The class is in great demand because there is a large number of individuals who are responsible for the deployment of voice-over-frame or voice-over-ATM technology.

Voice over IP
Cisco has announced a voice card for the 36xx routers (3620, 3640). It provides 2 or 4 analog voice circuits over IP (digital circuits are coming soon). This gives you up to 12 analog circuits in a 3640. If you need more, use the digital circuit capability. Also consider that at typical voice ratios of circuits to users, this might support many times more users, depending on your sites' calling patterns.

If your 2500 is running out of steam in some "power branch offices", this adds to the possible value of upgrading to a 3600 model. Cards for some other router models are in the works. Phones and faxes can plug directly into the 3600 card, or the key system or PBX can connect to it. A Java application is available to configure voice ports, dial plan, and manage the system. Reporting is also in the works.

In terms of signaling, the voice card does E&M, FXO, FXS:

• E&M: popular PBX to PBX or switch to switch trunking
  
• FXO: Foreign Exchange Office: looks like a phone or company to the PBX, allows off-premise connections
  
• FXS: Foreign Exchange Station: provides standard phone and fax connections

The point to the voice support in the 3600s is that the corporate IP network is made to appear as a trunk line to the PBXs.

When there's a call, a PBX can signal the connected router via the Q.931 message format (ISDN). This is passed to H.323 (which uses Q.931 for call signaling), and at the other end is passed as a line seizure to the PBX. The routers then forwards dial digits to the PBX. If the router can interact with the signaling, it opens the door for additional functionality. (Note that Cisco recently bought LightSpeed International, Inc., adding to their phone signaling capabilities).

The expected applications are:

• Toll bypass: PBXs can route internal calls through 3600s and over the data network, instead of as a toll call.
  
• PBX extension: phones and faxes plugged directly into the 3600 card appear as extensions to a PBX connected to a central 3600. They can also dial directly into the PSTN.
  
• H.323 interoperability: desktop applications using H.323 can bypass the local loop, they can also be gatewayed to the corporate PBX.
  
• PSTN gateways: users with desktop applications (EtherPhone) can use the 3600 as a gateway to the PSTN (Public Switched Telephone Network). This can be done internally or by an ISP.
  
• Offloading voice mail from a PBX to desktop PCs. (Admin., Reliability).

One (somewhat future) attraction of this is that IP networks are connectionless, so if E.164-style telephony addresses are mapped to IP addresses, dynamic IP routing rather than static call tables can be used. This has the potential to simplify managing a voice network and reduce costs.

The 3600 cards are intended to have VoFR and VoATM capabilities in the future. (And the MC3810 is acquiring VoIP capabilities).

Design Issues
Just a couple of important things, since space is short and design for low delay is a complex subject.

The MC3810 requires one at each end. Interoperability with an IGX card is intended; the card may be available by the time this article appears. The IGX card would allow higher density at the central site, aggregating the voice connections to all the remote sites with 3810s.

I noted elsewhere that delay is the biggest factor. That's been repeated since it's important. This is the current real barrier to corporate voice over the Internet. Fax relay traffic is a fairly easy win, though: fax delivery doesn't have to be real-time. Be careful here too, I gather that fax machines have tight end-to-end timing, so that you do need low delay for real-time fax delivery.

Some ideas on how to bring the end-to-end delay down:

• better PC drivers, Win95 direct drivers
  
• V.32, V.34 bypass: ISDN, Cable or xDSL
  
• better (low delay) networks: RSVP & ForeSight
  
• low delay Codecs

Voice/data articles*

• Topic: http://www.cisco.com/warp/customer/803/4.html
  
• Topic: http://www.cisco.com/warp/customer/728/3800/2310_ds.htm
  
• Topic: http://www.cisco.com/warp/customer/146/january98/16.html
  
• Topic: http://www.cisco.com/warp/ customer/146/january98/21.html
  
• Topic: http://www.cisco.com/warp/ customer/733/7200/alien_an.htm

• Selsius-Phone: full-featured phones that plug directly into the LAN
  
• Selsius-CallManager: performs PBX call control on the data network
  
• Selsius-Access: allows the IP PBX to connect to the Public Switched Telephone Network

The Call Manager software works with existing telephony systems and also can supply full PBX functionality on its own. It runs on an NT server using TCP/IP and provides the PBX features listed above, plus call park, calling party id to the Selsius-Phone. There is also a Unified Messaging Interface providing connectivity to voice mail and interactive voice response systems. It works with any H.323 based client. Because CallManager is based on IP, it can be deployed as a single IP PBX with geographically dispersed users.

The Selsius-Phone is a PBX-like phone that plugs directly into an Ethernet jack. It does not require a paired PC. The phones use DHCP to make phone setup "virtually automatic". Because of this, the phones can be easily moved and plugged in anywhere on the IP network with no configuration. Models include 12 or 30 programmable buttons, speakerphone, and display, and use G.711 and G.723 audio compression. The phones are Microsoft NetMeeting enabled, so application sharing and videoconferencing are available by pressing a button on the phone. Ringing and ring volume adjustments are available, configured via Web browser.

There is also a Selsius-VirtualPhone, a software-only version.

* Access for registered users only.