ZyXEL Communications Prestige P2000W V2 User manual

P-2000W V2
VoIP wireless phone
Troubleshooting Guide
May 2005

P2000W V2 Troubleshooting Guide
Table of Contents iii
Customer Support
Please have the following information ready when you contact customer support.
•Product model and serial number.
•Firmware version of your device.
•Brief description of the problem and the steps you took to solve it.
•Call server type and vendor.
•Packet trace log as instructed on the troubleshooting in this guide.
Table of Contents
Introduction............................................................................................................................................Page 4
Frequently asked question....................................................................................................................Page 4
VoIP troubleshooting.............................................................................................................................Page 9
The VoIP Challenge ...............................................................................................................................Page 14

P2000W V2 Troubleshooting Guide
All contents copyright (c) 2005 ZyXEL Communications Corporation.
Introduction
The Prestige P2000W V2 is a SIP-based VoIP wireless phone. Sending voice signals
over the Internet is called Voice over IP or VoIP. Session Initiated Protocol (SIP) is
an internationally recognized standard for implementing SIP.
The Prestige allows you to receive Voice over IP calls. You can call any landline or
mobile telephones as well as IP telephones. Calls received from IP telephone work
exactly as you would expect from traditional telephone service.
Although SIP is an internationally recognized standard there are still possibility of
interoperability issue occur with ITSP due to vendor’s different interpretation of the
SIP standard. Therefore we suggest an interoperability test with the call server is
conducted prior to device deployment.
Below are some frequently asked question which can help a better understand what is
VoIP.
Frequently Asked Question
What is VoIP?
How does VoIP work?
Why use IP for voice?
What is the relationship between codec and VoIP?
What advantage does VoIP can provide?
What is the difference between H.323 and SIP?
Can H.323 and SIP interoperate with one another?
What do I need in order to use SIP?
What is voice quality?
How are voice quality normally rated?
What is codec?
What is the relation of codec and VoIP?
Which codec should I choose?
I have press load default setting from Menu->Setting->Load Default option
from the phone cursor but the device seems not fully reset?
How do I reset the device to factory default?
What is the P2000W V2 supported Codec?
What is the P2000W V2 voice codec packet size and packet interval
relationship?
Can P2000W V2 support outbound proxy?
Can P2000W V2 support STUN?

P2000W V2 Troubleshooting Guide
All contents copyright (c) 2005 ZyXEL Communications Corporation.
1. What is VoIP?
Voice over IP is an emerging technology based on open standards of IEEE,
fundamentally the Internet Protocol that allows voice data to travel across the
Internet. There are many methods to use this technology, the most common
and well known are SIP, and H.323.
2. How does VoIP work?
Basically VoIP is a technique to send voice information in digital form in
discrete packets over digital network rather than by using traditional circuit
switch (PSTN). To do so we will need an analog to digital converter on
sender side to translate the voice (analog signal) to digital than transmit it, and
on the receiver end it will also need an analog to digital converter to covert the
digital signal back to analog to the person being called can heard the voice.
3. Why use IP for voice?
Traditionally telephony carrier use circuit switching for carrying voice traffic. As
circuit switching is designed to carry voice and it does it very well. Than why use IP
for voice? As broadband booms, and technology evolve. People now want to
communicate through various way not just voice such as email, instant messaging,
video and so on. Traditional telephony can not evolve as quickly as the demand and
develop new feature on circuit switch takes much time and money. IP is an already
exist standard and many type of service already runs on IP, by using IP as a platform
integrate service is now possible and low cost where traditional circuit may take long
time to achieve.
4. What is the relationship between codec and VoIP?
In order to transfer voice (analog signal) over IP it first needs to be
digitized. Codec is a technique to digitize analog signal to digital and vice
versa. There are various speech codec available and can be used with VoIP
each with its advantage and disadvantage.
5. What advantage does VoIP can provide?
The advantage of VoIP is it can provide advance services such as joining e-
mail, instant messaging, video, voice mail all together. Where current circuit
switching (PSTN) can not.

P2000W V2 Troubleshooting Guide
All contents copyright (c) 2005 ZyXEL Communications Corporation.
6. What is the difference between H.323 and SIP?
H.323 and SIP are proposed by different group
Session Initiation Protocol (SIP) is a standard introduced by the Internet
Engineering Task Force in 1999 to carry voice over IP. Since it was created
by the IETF, it approaches voice and multimedia from the Internet, or IP,
perspective of view. Where as H.323 emerged around 1996, and as an
International Telecommunication Union standard it was designed from a
telecommunications perspective. Both standards have the same objective - to
enable voice and multimedia convergence with IP protocols.
7. Can H.323 and SIP interoperate with one another?
In interoperability between the two, the industry is making slow but sure
progress. Interoperability must first happen between vendor implementations
of the same protocol (SIP-to-SIP and H.323-to-H.323) and then between
protocols. Currently in order for SIP client to talk to H.323 client the ITSP
must have a trunking gateway act as a translator between the two protocols
without the truncking gateway the two protocols are not able to communicate
to one another.
8. What do I need in order to use SIP?
The minimum required to use VoIP is as follow.
1. A high-speed Internet connection. This can be a cable modem, or a high-
speed network services such as ISDN, DSL or a T-1 link. The need of
the bandwidth required will depend on the amount of telephone traffic will
be in your network.
2. A PC with VoIP software installed or a hardware VoIP box such as ATA
or device like P2000W V2.
3. An account with a VoIP provider such as an ITSP. The account can be
configured to recognize your calls automatically, or you can require the
users to enter their unique account numbers issued.
9. What is voice quality?
Voice quality is how well a person can hear the voice on the opposite end.
10. How are voice quality normally rated?

P2000W V2 Troubleshooting Guide
All contents copyright (c) 2005 ZyXEL Communications Corporation.
Voice quality is most commonly rated through a voice quality metric called the
Mean Opinion Score (MOS) which is recommendation by ITU-T. The MOS
is a 5 point scale where 5 represent excellent voice quality and 1 represent
bad voice quality.
11. What is codec?
Codec is an algorithm which converts analog signal into digital signal and vice
versa. There are three main types of waveform codec, source codec, and
hybrid codec. Each consume different amount of bandwidth and provide
different voice quality level.
12. What is the relation of codec and VoIP?
As VoIP is general term send voice information in digital form in discrete
packets over digital network and this digital network is public network, thus
there maybe other packet such data packet uses network at the same
time. The codec choose is related to how much bandwidth voice packet will
consume. In bandwidth wise aspect the smaller amount of bandwidth used
the better. But in voice aspect the higher quality the better.
13. Which codec should I choose?
As which codec choose is depending on what codec is supported on both end
of the VoIP host. Generally a codec with low bandwidth consumption and
high voice quality is a good codec.
14. I have press load default setting from Menu->Setting->Load
Default option from the phone cursor but the device seems
not fully resetted?
'Load default' is not reset all of the settings. If you press the 'load default', it
will reset the Time format, Key lock, Melody, Ring volume, Banner and
Vibrator setting to default.
15. How do I reset the device to factory default?
To reset P2000W V2 to default including username/password. Please follow
the step as below.

P2000W V2 Troubleshooting Guide
All contents copyright (c) 2005 ZyXEL Communications Corporation.
1. Power off the device.
2. Keep hold down the Power and Left-Selector button until you
see <<Load Default>> <<Factory>> words in the LCD screen than release it.
3. Now all setting has been reset to default.
16. What is the P2000W V2 supported Codec?
P2000W V2 supported codec are
G.729 voice codec
G.711u-law voice codec
G.711a-law voice codec
17. What is the P2000W V2 voice codec packet
size and packet interval relationship?
P2000W V2 Voice Frame per Packet V.S.
Frame Size G.729 codec
Small Medium Large
RTP Payload 20 byte 32 byte 60 byte
Voice Packet Size 74 byte 86 byte 108 byte
Packet Interval 20 ms 30 ms 60 ms
G.711 u/a-law codec
Small Medium Large
RTP Payload 240 byte X 480 byte
Voice Packet Size 294 byte X 588 byte
Packet Interval 30 ms X 60ms
Note:
1. Voice packet size = Ethernet header (14byte) +IP header (20byte) +UDP header (8byte)
+RTP header (12byte) +RTP payload. Therefore voice packet size=54byte+RTP payload
size.
18. Can P2000W V2 support outbound proxy?
P2000W V2 can support outbound proxy.

P2000W V2 Troubleshooting Guide
All contents copyright (c) 2005 ZyXEL Communications Corporation.
19. Can P2000W V2 support STUN?
Yes, P2000W V2 can support STUN.
VoIP Troubleshooting
Is it a hardware problem?
How can I check if problem is cause by NAT or not?
If I've confirmed it is not an NAT. What should I do?
Unable to register with the SIP server?
I can register but can not establish a call?
I can make a call but the voice only goes one way not bothway?
I can receive a call but the voice only goes one way not bothway?
If all the about have been tried, but register still fail or call still have
problem to be made behind the NAT what should I do?
The support tells me, I may have an interoperabiltiy issue with my ITSP
call server. Packet trace is requested by support to analyze the
problem. Should I setup the environment and how do I trace the packet?
1. Is it a hardware problem?
If following symptoms happen on the P2000W V2
•LCD display does not show “Initializing please wait message” when
turn on
1. Make sure the battery of the phone is fully charged.
2. You are using the charger base and power adapter to charge and the
power adapter are plugged in to an appropriate power source.
3. Check that the power source is turned on. If the problem persists, you may
have a hardware problem. In this case, you should contact your local
vendor.
•The P2000W V2 sometimes shut down automatically
Check if the battery on P2000W V2 is too low. Check that the P2000W V2 is
fully charged, if it is fully charge but standby time relatively low the battery
may need to be replaced.

P2000W V2 Troubleshooting Guide
All contents copyright (c) 2005 ZyXEL Communications Corporation.
2. How can I check if problem is cause by NAT or not?
There may not be an easy or quick answer to this question. One of the best
way to see if the problem is cause by NAT or not is to put P2000W V2 on
public network, (With public IP and no NAT) see if it work under such
condition.
If you it can work on public network but can not work behind NAT than very
likely your NAT gateway is not supporting SIP pass-through. In such
condition you will need to consult your NAT router vendor for resolution.
3. If I've confirmed it is an NAT problem. What should I do?
If you are certain or confirmed the problem is caused by NAT you need to
check the following.
1. Check if your NAT router vendor supports SIP pass-through.
2. If the NAT router does not support SIP pass-through, you need to consult
your ITSP support if they support SIP outbound proxy. Since different NAT
router vendor implement NAT differently. Typically ITSP may provide SIP
outbound proxy to resolve NAT pass-through issues as sip proxy does not
require the SIP client to do any modification as the server will modify the SIP
message content according to the your WAN IP it sees. If your ITSP
supports outbound proxy you need to obtain the configuration and configured
the outbound proxy location on P2000W V2.
3. As mentioned some NAT may not be SIP friendly. ITSP may support
STUN instead of SIP outbound proxy to resolve this issue. To use STUN sip
client must support STUN client. How STUN work is when the user register
the server will tell SIP client what the WAN IP the server sees from the SIP
client and SIP client will use this IP as SIP content to identify its location.
4. Unable to register with the SIP server?
If you are unable to register with SIP server.
1. Make sure the Internet is reachable and the SIP register server is
reachable. If your register server uses domain name make sure DNS
name can be resolved.
2. Make sure the SIP account is correct and the password is correct.
3. Check if there is NAT router before it. If there are NAT router before it
follow troubleshooting step above to verify if NAT is the cause of the
problem.

P2000W V2 Troubleshooting Guide
All contents copyright (c) 2005 ZyXEL Communications Corporation.
5. I can register but can not establish a call?
If you can register to server but can not make a call very likely there is NAT
router before it, please see NAT related question above for tips to
troubleshoot.
6. I can make a call but the voice only goes one way not
bothway?
If you can register to server but can make a call signal establishment but the
voice only goes one way very likely there are NAT router before it, please see
NAT related question above to tips to troubleshoot.
7. I can receive a call but the voice only goes one way not
bothway?
If you can register to server but can only make out going call but can not
receive incoming calls or the incoming call signal establishment can be made
but voice only goes one way very likely there is NAT router before it, please
see NAT related question above for tips to troubleshoot.
8. If all the about have been tried, but register still fail or call
still have problem to be made behind the NAT what should I
do?
In such case, please contact your local vendor for support. If they can't help
out the problem they will escalate your problem to ZyXEL tech center. To
report a problem please prepared below info.
1. Serial number of the device.
2. SIP Call server type and vendor.
3. Your device firmware version and romfile with password.
4. Detail information what you have tried to resolve the problem.

P2000W V2 Troubleshooting Guide
All contents copyright (c) 2005 ZyXEL Communications Corporation.
9. The support tells me, I may have an interoperability issue
with my ITSP call server. Packet trace is requested by
support to analyze the problem. How should I setup the
environment and how do I trace the packet?
In such as you will need to setup your environment similar as scenario A & B
to perform packet trace. Make a failed trace and make a successful trace by
setting up scenario A & B and send it to support for problem analysis.
To perform packet trace the equipment required is as follow
1. PC installed with packet sniffer such "Ethereal" to make the packet
capture. Ethereal is a freeware and can be downloaded from
www.ethereal.com to perform the packet capturing.
2. Hub and Ethernet cable.
3. Follow the scenario below to capture a working and a none-working trace
with Ethereal and send it to support.
Scenario A
1. Connect the hub between the P2000W V2 and the front end NAT router
with Ethernet cable.
2. Connect the PC with packet sniffer installed to the hub.

P2000W V2 Troubleshooting Guide
All contents copyright (c) 2005 ZyXEL Communications Corporation.
3. Start packet sniffing software and start packet capturing.
4. Start P2000W V2 and make a complete call from registering to calling a
party until problem appears.
5. Stop capturing and save the captured packet in file.
Scenario B
1. Connect the hub between the P2000W V2 and the front end CPE with
Ethernet cable (Make sure P2000W V2 is with public IP).
2. Connect the PC with packet sniffer installed to the hub.
3. Start packet sniffing software and start packet capturing.
4. Start P2000W V2 and make a complete call from registering to calling a
party.
5. Stop capturing once call is established and voice can transferred both way
successfully.
6. Save the captured packet in file.

P2000W V2 Troubleshooting Guide
All contents copyright (c) 2005 ZyXEL Communications Corporation.
The corrected working capture should include starting of the call (The SIP
invite) to call pickup (The SIP ACK). An example of successful call capture is
listed below for your reference.
Once both of the above capture is completed, send both captured file to
support to analysis the cause.
The VoIP challenge
SIP is a session initiation protocol which is an application-layer control (signaling)
protocol for creating, modifying, and terminating sessions with one or more
participants. These sessions include Internet telephone calls, multimedia distribution,
and multimedia conferences.
For SIP to successfully massively deploy there are several challenge of the real
environment it must able to work with. The most common faced problem is NAT
pass-through issue and SIP client and call server interoperability issues.
When SIP protocol is proposed it self does not consider network environment and
other technology that maybe in the deployed network. Since IPv4 IP is insufficient
and will soon run out. Thus NAT is very commonly used. Unfortunately from field
test many vendor’s NAT are not SIP friendly.

P2000W V2 Troubleshooting Guide
All contents copyright (c) 2005 ZyXEL Communications Corporation.
Fortunately, there are already ways to workaround the NAT for SIP. The most
commonly used way is listed below.
SIP pass-through support on NAT router (require router vendor implement it).
STUN.
Outbound Proxy.
There are advantage and disadvantage of each technology.
For example SIP pass-through on NAT router this requires router vendor
implementation. From carrier point of view this does not seem feasible as not every
NAT router vendor implement this on their router and they do not have control of
which router their customer is using.
STUN is a simple traversal of user datagram protocol. It helps to traverse UDP
through NAT. STUN allows applications to discover the presence and type of NATs
between the host and public Internet. It also provides the ability for application to
determine the public IP allocated to the host by NAT. STUN works with many
Existing NATs, and does not require special handling from NAT router. As a result it
can make SIP pass-through NAT. Please refer to RFC 3489 for more details.
The outbound proxy is a normal SIP proxy. You configure your client, the phone or
software, to use the proxy for all SIP sessions, just like when you configure your Web
browser to use a Web proxy for all Web transactions. In some cases, the outbound
proxy is placed alongside the firewall and is the only way to let SIP traffic pass from
the internal network to the Internet. For more detail you can refer to RFC 3261 for
details.