VoIP Info

 

http://www.voip-info.org is your best friend

As you know, business telephone systems have historically offered limited (or no) capability in terms of supporting a remote home-office telecommuter. Additionally, we have grown used to excellent sound quality at the office. With VoIP capable PBX systems like PBXtra, the paradigm is changing. New capabilities are now possible that let us do more, and save money in the process.

PBXtra offers a feature that allows remote telecommuters to participate on the main PBXtra system through the internet just as if they are in the office. The sound quality of a remote telecommuter connection may vary depending on a number of factors. Remote telecommuter connections require solid Internet connectivity between the main office location and the remote location.

Also, telephone companies are beginning to offer SIP and IAX2 based VoIP toll services that allow you to replace some (or all) of your phone lines with digital services that allow you to send and receive calls from any telephone. Planning the right network for using this sort of a service is essential to ensure good performance.

This document includes a high-level overview of these subjects, including a technical description of why call quality is not always perfect. I’ll also offer some suggestions for how to plan for the best possible telecommuter configuration.

In A Nutshell

Local

  • PBXtra needs to have a unique static IP on your network OR a DHCP reservation

  • PBXtra needs to be on the same side of the NAT as the bulk of your phones

  • PBXtra needs to be the primary DNS server on your network

    • In the PBXtra's internal IP address settings, it should refer to an external DNS server.

  • Your phones can be DHCP clients or assume static IP's, as long as there are no conflicts of IP addresses

 

Remote

  • Your phones must have the 'x' in the domain name for registration as described in the [Setting Up Remote Phones] article

    • The 'x' in the registration name is not necessary if you are using a VPN

  • Both the phones and the PBXtra must have clean unfettered access over port 5060

  • Only one phone per NAT device! Your phones must appear to the PBXtra on unique IP addresses

    • Multiple phones at a remote site can best be accomplished with a VPN

 

About the SIP VoIP Protocol

PBXtra’s IP phones use a VoIP protocol called SIP. This is a UDP/IP based protocol that loosely resembles FTP from a protocol perspective. Yes, FTP uses TCP, and SIP uses UDP, but they use a similar approach with respect to control and streaming of data. SIP uses a control channel used to set up calls (UDP/5060), and a dynamically negotiated port number to transmit the actual call audio. This “multimedia” path that carries the audio is known as RTP, and is also a UDP transmission. Why UDP? Because UDP has low overhead compared to TCP, and lends itself well to this sort of a voice transmission.

 

Ideal Network Conditions

Remote IP phones that use the Internet to connect to your main location require a high performance congestion free connection between the remote location, and the main PBXtra location. The ideal network conditions for a remote telecommuter are:

  1. Low Latency. This means that the round trip time for a packet between the two locations is less than 100ms on average.

  2. Jitter Free.

    Jitter is the variance in latency between packets. For example, if you measured the network performance using “PING” and you see 90ms, 90ms, 91ms, 89ms… then you have a jitter free network. If it measures 20ms 130ms 60ms 35ms, then you have jitter, and this will adversely impact sound quality because packets will arrive out of order.

  3. No Packet Loss. If an Internet network is congested, sometimes the data is discarded, resulting in a partial delivery of the data. If this happens, portions of your call will be stuttered or silenced. If there is any packet loss at all between your locations, call quality will suffer.

 

Web works great, but sound quality is poor. Why?

The web uses the HTTP protocol over TCP/IP, which is not sensitive to Latency, Jitter, and Packet Loss like VoIP is with SIP over UDP/IP. The same is true for email, using IMAP, POP, and SMTP protocols. Web and email use TCP/IP which will automatically retransmit any missing data. A connection that appears to work well for email and web browsing may be totally unsuitable for SIP.

 

Where to look for network problems

Firewalls on either end of the remote connection can introduce jitter. Jitter can also be caused by a saturated internet connection. Saturation can occur by sending large amounts of other data (Web browsing, Email Downloads, etc.) while using the IP phone. If your router or firewall has QoS features, consider using them to give priority to your VoIP transmissions. See Also: How do I use QoS on my network?. If the Internet is congested anywhere between the two locations, it will likely result in reduced audio quality. Congestion can occur:

  • At the main office location

    • Typically caused by heavy use of Web and email

    • Sometimes caused by networked “worm” computer viruses.

    • Can be solved using QoS equipment and configuration.

  • At the remote location

    • Typically caused by using the web or email while on a call.

    • Sometimes caused by networked “worm” computer viruses.

    • Sometimes caused by cheap broadband modems not designed for VoIP use.

    • Can be solved using QoS equipment and configuration.

  • Between the two locations on the internet

    • You cannot control this.

    • Try to use the same internet provider at both locations to keep this to a minimum.

 

About DSL/Cable Internet Access

Most remote telecommuters have DSL or Cable Modem service at home. These connections can work very well for using a PBXtra phone as a remote telecommuter, provided that the connection is congestion free. Most of these connections use very inexpensive modems (aka: CPE). Typically these modems cost under $50 to manufacture, and many corners have been cut in order to make them cheap. One common problem is that they will share a single packet buffer for both inbound and outbound packets. That works fine for web browsing and email, but will destroy VoIP performance when you begin sharing the connection between VoIP, and other protocols. For example, a single VoIP call will sound just fine until your email program downloads a message. All of a sudden your VoIP connection sounds bad. This can be prevented. See suggestions below.

T1 connections are almost always serviced by a router that has superior performance to any Cable/DSL modem. That’s why T1 lines work so much better than DSL for remote telecommuters. Many T1 routers also include QoS capabilities that can be activated to further enhance VoIP performance. In addition, internet providers typically offer a higher level of performance to their T1 customers because of lower oversubscription rates. It does cost more for T1 server, but it’s worth every penny if you plan to run VoIP over the internet.

 

Formula for the best remote telecommuter Experience

  1. Use T1 internet access at the main location, not DSL or Cable. It’s worth the additional expense in order to ensure good, steady performance at your main location.

  2. If your routers and/or firewalls support QoS features, activate them. Give priority to the SIP and RTP protocols. Consider replacing equipment that lacks VoIP-aware QoS features. See Also: How do I use QoS on my network?

  3. Consider using one of our Suggested Routers with QoS on both ends of your connection.

  4. If your QoS solution allows you to limit total bandwidth, set the limit to slightly less than the line speed of your internet connection. Use a DSL line speed test to determine where you should set your limits. Setting it about 5-10 Kb below your maximum speed will keep the packet buffers from filling up on your DSL/Cable modem. This will yield better overall performance.

  5. Consider having two internet connections… one for your existing data application, and one for your VOIP phone and PBXtra servers. You can use this approach in your main location, as well as your remote locations. If you use this approach, you may not need any QoS capable equipment.

  6. If possible, connect your main office and your remote office using the same internet provider. Usually performance on the same provider’s network is superior to the performance when traffic needs to traverse multiple internet backbone networks.

  7. If possible, remove NAT devices between the PBXtra system, and the remote telecommuters.

  8. If you must use a NAT configuration, consider using a “DMZ Host/Server” configuration rather than port forwarding. This uses less CPU power in the router/firewall and yields optimal performance.

    1. At the main location, the setting will forward all unknown packets to your PBXtra server.

    2. At the remote locations, the setting will forward all unknown incoming packets to the IP Phone.

    3. Reserve the phone’s IP address in DHCP or give the phone a static IP Address on your private network in the remote location so the IP Address does not change. If you use a static IP Address, pick one outside of your dynamic DHCP IP Address range.

  9. For mission critical remote employees, consider using a fractional T1 internet service at the remote office instead of a Cable/DSL connection.

 

Formula for the best VoIP toll service

If you plan to use a VoIP service provider for incoming or outgoing telephone calls to/from the PSTN (Public Switched Telephone Network) in order to reduce your long distance charges, be careful. It’s easy to put yourself in a situation where all of your call quality suffers as a result of an inadequate network configuration. Usually we find that using traditional PSTN connections like a T1/PRI are cost neutral to a properly engineered VoIP solution. Look at all your options carefully before deciding to use a pure VoIP solution.

  1. Select a good VoIP Provider

    1. Find one with a good reputation for quality and customer service.

    2. Make sure they have a local PSTN media gateway in your area. For example, if your office is in Seattle, and the VoIP provider only has facilities in Miami, you should probably not select that provider because all of your calls will need to go over the internet from Seattle to Miami before they reach the PSTN. The shorter the path, the better.

    3. Be sure your VoIP provider is E911 compliant, and that they can offer you routing to E911 facilities for ALL of your phone numbers. If they are not, then install an ordinary phone line in your PBXtra system to be reserved for calling 911 in the event of an emergency.

  2. Do not attempt to use Cable/DSL internet service for your main office location. We have learned from experience that this has less than a 50% success rate. Don’t waste your time, really.

  3. Consider a VoIP service provider that offers dedicated T1 network access from your main office location directly to their VOIP network. Try not to rely on the internet to reach your VoIP provider.

  4. Be sure to use a QoS solution if you plan to run VoIP and web/email on the same connection.

  5. If you must use a NAT configuration, consider using a “DMZ Host/Server” configuration rather than port forwarding. This uses less CPU power in the router/firewall and yields optimal performance. This setting will forward all unknown incoming packets to your PBXtra server.

  6. Do not attempt to send or receive any faxes using your VoIP service. Your PBXtra system does not have any internet fax support (as of 11/2005).

As you can see, using VoIP can add additional complexity to the planning process when setting up your system for remote telecommuters or VoIP toll service for your long distance calls. Spending the time up front to plan and budget for the right networking facilities, equipment, and service will save you many headaches down the road when it comes to call quality concerns.

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