Echo is a problem that is exacerbated by latency and the use of hybrid interfacesthose that provide VoIP servers with a way of using legacy phone lines. Avoid latency and legacy phone lines, and you solve the echo problem. If you can't avoid them, then use high-quality hybrid interfaces like those available for Digium's TDM400P or Intel Dialogic cards. Asterisk users can also enable aggressive echo suppression, as in Chapter 9
SIP registrations and RTP media streams will fail through a NAT firewall unless a DMZ-based SIP proxy is used. SIP servers need non-NATted access to both the Internet and the private IP network, so that they can serve as an application proxy for SIP endpoints. DMZ provides this non-NATted access
If an ATA for a TSP such as Vonage or Packet8 works fine for a few days and then suddenly fails, requiring a reboot to remedy the failure, it's likely that your broadband router has obtained a new IP address and the ATA doesn't know what it is. Rebooting the ATA allows it to "sense" the new address, and reestablish its path to the TSP's server
Callers sound robotic when one too many codecs or sampling conversions occurs. Going from analog to G.729A to analog to G.711 ALaw will cause quite a robotic-sounding call, so avoid excessive transcoding and re-sampling
Your private-premises VoIP system must draw power from your own facilities, not from the phone company's. Make sure there's adequate backup power, not just for your servers, but for your endpoints. PoE is a good solution to this problem (see Chapter 8 for a review of PoE)
Use a regular backup routine to keep several of the most recent revisions of your dial-plan and voice peering configurations on hand, in case of a natural or administrative disaster. Keep copies of the backups off-site if possible
