Vodavi Phone Faqs

V O D A V I C O M M U N I C A T I O N S

Frequently Asked Questions: Network Issues

By: Dean Altman MCSE, CCNA and Cindy Domangue MCSE, MCDBA, N+

Glossary

Question #1: How do you calculate VoIP Bandwidth?

Question #2: What is the difference between ToS (Type of Service) and DSCP (Differentiated Service Code Point)?

Question #3: How do I program voice networking over a LAN/WAN between Telenium systems?

Question #4: How do I program voice networking with Centralized Out-Dialing on the Telenium?

Question #5: How can I Prioritize voice traffic on my Cisco router?

Question #6: How do I test SMTP ports on a mail server with a telnet connection?

Question #7: What are Jitter, Latency and Packet Loss? And, what can I do about them?

Question #8: What is a NAT? And, how can I set-up and use one with my VoIP equipment?

Question #9: Why does my PathfinderIP not play any prompts?

Question #10: What are the most common errors when programming a Telenium?

Question #11: How can I wipe the file system clean on the Telenium and start over?

Question #12: What is the difference between VPN pass-through and VPN Tunnel?

Question #13: How do I upgrade the VoIP Card Software to 3.1d?

Question #14: How do I set-up PathfinderIP for Multiple Tenants?

Question #15: How do I set up an RSGM at home on a Telenium IP System?

Question #16: How do I delay the PathfinderIP SystemManager.exe so my services can start?

Question #17: How do I program IP phones on a VoIP card?

Question #18: How do I choose IP addresses?

Question #19: How do I program an Adtran 3200 router for a point to point T-1 to network an XTS?

Question #20: How do I connect to an Analog Pathfinder VM with a Telenium?

Question #21: How do I upgrade a T-1 Module on the Telenium?

Question #22: How do I set up an RSGM Module on an XTS VoIP Card?

Question #23: How do I set up centralized out-dialing in LCR on systems networked by VoIP cards?

Question #1: How do you calculate VoIP Bandwidth?

For basic VoIP bandwidth calculation begin with the codec type. G.711 is utilized for the best quality of voice and for proper transmission of DTMF tones. This codec does not compress the voice and utilizes a Ў§full 64KbpsЎЁ of voice bandwidth. The value is generated using 125 microsecond delays between voice samples (itЎ¦s a long mathematical formulaЎKsuffice to say itЎ¦s 64 Kbps for G.711, 8 Kbps for G.729 and 6.3 Kbps for G.723.1). However, additional considerations change the amount of bandwidth actually used.

The amount of milliseconds (of voice) in a packet determines the number of packets required per second. If you have 50 milliseconds of voice per packet, you need 20 packets per second. The reverse is true if you want 20 milliseconds of voice per packet; you need 50 packets per second.

Each packet acquires various amounts of information depending on the network type in the packet header. For example there are approximately 40 bytes in a header for Ethernet packets. Each time a packet is sent, the header must be attached increasing the size of the packet and therefore