1120 VoIp Phone Issues 3

[jimmers]

Customer has a BCM 50 with VoIP 1120 phones at the main location and a remote location. They have a dedicated VPN tunnel to the main site for the remote location. The problem is that the phones are constantly dropping and when they do work, the sound quality is terrible. This happens at both locations.

We ran a 2 hour ping test from a field office computer to the BCM. They averaged under 100 ms but did have ...

about a 1% packet loss and
response times (occasionally) of up to 3,000 ms!

Obviously, they won't work very well if the network is essentially dead for 3 seconds.
Does anyone know any specification for what your IP phones require? I've tried changing codecs on the phones but it doesnt help.

 

[Dasheen]

do the payloads match? do you have a voice vlan for the sets and a separate vlan for the data? Did u do a network assessment before adding ip sets to the network?

 

[jimmers]

Could you elaborate a bit more on the payload size? As far as I know the vlan for the phones is different then the data. Unfortunately a network assessment wasn't done before they were install, but I won't get into that!
If it matters the phones are running off a POE switch.

 

[gwebster]

There are three network characteristics that affect voice quality:
- delay (how long it takes packets to traverse the network)
- packet loss (how many packets get lost along the way)
- jitter (variation in how long packets traverse the network)

Your delay and packet loss are not bad but your jitter appears to be huge. Unfortunately, voice requires packets to arrive at a constant rate with a fixed delay between those packets. If the next packet is not received in time, it has to be dropped and considered as lost. If jitter is high, whole groups of packets can be late or lost and result in garbled speech. All netwokrs have some jitter and built-in "jitter buffers" handle those variations.

 

[jimmers]

How can I determine if the payloads match? What is an acceptable response time?

 

[gwebster]

Perceived voice quality is a function of codec, packet loss, and delay (including change in response time due to jitter).

Although not a BCM document, grab a copy of the CS 1000 "Converging the Data Network with VoIP Fundamentals" NTP from

http://www.nortel.com/support.

There is a good writeup of how these various factors come into play and a table which gives an estimated voice quality based upon the three variables.

 

[gm85]

as gwebster was mentioning, VoIP is a very picky protocol. Unlike FTP, POP3, HTTP, etc, if the conditions aren't met (most importantly jitter and latency) VoIP will not work well.

There are two protocols that are used with VoIP setups: a voice protocol and a signaling protocol. The voice protocol is the most obvious (and carrys the voice conversation). The signaling protocol is what controls your phone and tells it to blink, ring (as well as tells the BCM what you are doing). If the voice protocol runs into problems, you get call quality issues. if the signaling protocol runs into problems, the phones will become unresponsive and reboot.

My guess is that your internet connections are overloaded. Do you have QoS in place on your edge routers? Unless you have a gigabit connection, QoS is absolutely necesary to ensure that the voice/signaling packets can get through, regardless of how saturated your internet connection is.

 

[jimmers]

QOS is enabled, and set to the default of level two. Should I set the QOS at the highest level of 4? What is the minimum that the phones need?