VoIP FAQ
What is Voice over IP?
P-2024 Support Notes
Voice over IP (VoIP) is an emerging technology based on the open IEEE standards. VoIP refers to the transmission of voice data over the Internet. Various protocols are available for voice transport. The most commonly used are SIP and H.323.
Voice over IP (VoIP) is an emerging technology based on open IEEE standards. VoIP refers to the transmission of voice data over the Internet. Various protocols are available for voice transport. The most commonly used are
SIP and H.323.
How does Voice over IP work?
In VoIP, voice data is sent digitally in discrete packets through the Internet, not through the traditional circuit switch of PSTN. To do so, an analog-to-digital converter is required at sender side to translate voice (analog signal) to digital signal before transmission. At the receiver end, an analog-to-digital converter converts the digital signal back to analog so the voice can be heard on the phone.
Why use VoIP?
Traditionally voice data is transmitted using circuit switching. Since circuit switching is designed to carry voice, it does it very well. However, as broadband networks become a mainstream for network access and technologies have evolved, we don't want to confine ourselves to just using text-based applications (such as e-mail, instant messaging, etc.) for communication over the Internet. Thus, the convenience of voice communication through the Internet has quickly become popular.
In addition, it would take a much longer time, more effort and money to implement new features using circuit switching. Since the IP technology is a standard and various applications are available, it is easier and more cost-effective to integrate new services
and applications using IP.
What is the relationship between codec and VoIP?
In order to send voice (analog signal) over IP, it first needs to be digitized. Codec is a technique used to digitize analog signal into digital signals and vice versa. There are various speech codec available for VoIP. Each codec has its advantages and disadvantages.
38
All contents copyright (c) 2006 ZyXEL Communications Corporation.
What advantage does Voice over IP provide?
P-2024 Support Notes
VoIP provides advanced integration of text, video and voice in emails. This cannot be done using traditional circuit switching (PSTN).
What is the difference between H.323 and SIP?
H.323 and SIP are proposed by different groups. 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. Whereas 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.
Can H.323 and SIP interoperate with each other?
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.
What is voice quality?
Voice quality is how well a person can hear the voice on the opposite end.
How are voice quality normally rated?
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.
What is codec?
Codec is an algorithm that converts analog signal into digital signal and vice versa. There are three codeec types: waveform, source, and hybrid codec. Each consume different amount of bandwidth and provide different voice quality.
39
All contents copyright (c) 2006 ZyXEL Communications Corporation.
What is the relationship between codec and VoIP?
P-2024 Support Notes
VoIP is the general term to refer to the sending of digitized voice information in discrete packets over public digital network (the Internet) where other data packets can be sent at the same time. A codec determines how much bandwidth voice packets will use. To save bandwidth usage, you would use as little bandwidth as possible at the cost of reduced voice quality.
What codec types does P-2024 support?
The P-2024 supports the following commonly used codecs.
• G.726
• G.729 a/b voice codec
• G.711u-law voice codec
• G.711a-law voice codec
• G.723.* (Option)
Which codec should I choose?
Choose a codec that is also supported on the remote VoIP host since both ends of the VoIP connection must use the same codec. In general, a codec with low bandwidth consumption and high voice quality is a good codec.
What do I need in order to use SIP?
The following lists the minimum requirement for running VoIP applications.
1. A high-speed Internet connection. You can connect to the Internet using a cable or DSL modem. Or subscribe to high-speed network services such as ISDN, DSL or T-1. The bandwidth requirement varies depending on the amount of traffic in your network.
2. A PC with VoIP software installed or an external VoIP gateway (such as an ATA or the P-2024 2602 VoIP station ATA).
3. An account from a VoIP services provider (such as an ITSP). The account can be configured to recognize your calls automatically, or you can require the users to enter their assigned unique account numbers.
40
All contents copyright (c) 2006 ZyXEL Communications Corporation.
I am unable to register to a SIP server
If you are unable to register to a SIP server, do the following.
P-2024 Support Notes
1. Make sure the Internet connection is up and that you are able to ping the SIP register server from the LAN behind the P-2024. If your register server uses a domain name, make sure DNS name can be resolved. If you are using a static WAN IP address, make sure the DNS server is configured correctly on your P-2024.
2. Make sure the SIP account is correct and the password is entered correctly. They may be case-sensitive.
3. Check if there is a NAT ATA install before the P-2024 which is a VoIP station gateway. It is NOT recommended that you install a NAT ATA in front of the P-2024 as this may cause unexpected problems. If you still want to install a NAT ATA, use a VoIP ATA (VoIP Analog Telephone Adapter), such as the P-2024
ATA series, instead.
I can register to the SIP server but cannot establish a call
If you are able to register to the SIP server but cannot make a call through the P-2024, it is very likely there a
NAT ATA or a firewall blocks the traffic.
It is NOT recommended that you install a NAT ATA in front of the P-2024 as this may cause unexpected problems. If you still want to install a NAT ATA, use a VoIP ATA (VoIP Analog Telephone Adapter), such as the P-2024 ATA series, instead.
I can make or receive a call but the voice traffic only goes one way, not both way
If you can register to a server and can only make an out- going call but cannot receive incoming calls or the incoming call signal establishment can be made but the voice traffic only goes one way, there is very likely a
NAT/firewall ATA installed before the P-2024. Refer to the NAT/firewall related questions for more information.
I have tried all the troubleshooting steps, but still cannot register to the SIP server. What should I do next?
In this case, contact your local service provider for support. If they cannot solve your problem, they will send your problem to the ZyXEL global technical support center. To help out the problem they will escalate your problem to ZyXEL tech center.
To help us solve your problem quickly, please prepared the following information.
41
All contents copyright (c) 2006 ZyXEL Communications Corporation.