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FAQ
General FAQ
- What is Voice over IP (VoIP)?
- The general goal behind Voice over IP, or IP Telephony, is to carry voice (telephone) traffic over the same network that carries data computer) traffic.
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- What's the difference between VoIP and voice over the Internet?
- Voice over IP system is NOT voice over the Internet. IP stands for Internet Protocol, which means that the voice is digitized into data packets so that it is compatible with being sent over data networks, including the Internet. However, VoIP only carries the call in IP format from your desk phone to the other phones.
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- How does VoIP work?
- Many years ago it was discovered that sending a signal to a remote destination could have be done also in a digital fashion: before sending it we have to digitalize it with an ADC (analog to digital converter), transmit it, and at the end transform it again in analog format with DAC (digital to analog converter) to use it. VoIP works like that, digitalizing
voip How Works
voice in data packets, sending them and reconverting them in voice at destination. Digital format can be better controlled:one can compress it, route it, convert it to a new better format, and so on; also digital signal is more noise tolerant than the analog one. TCP/IP networks are made of IP packets containing a header (to control communication) and a payload to transport data: VoIP use it to go across the network and come to destination.
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- What are the advantages of using VoIP than PSTN?
- When you are using PSTN line, you typically pay for time used to a PSTN line manager company: more time you stay at phone and more you'll pay. In addition you couldn't talk with other that one person at a time. In opposite with VoIP mechanism you can talk all the time with every person you want (the needed is that other person is also connected to Internet at the same time), as far as you want (money independent) and, in addition, you can talk with many people at the same time. If you're still not persuaded you can consider that, at the same time, you can exchange data with people are you talking with, sending images, graphs and videos.
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- What is SIP?
- The Session Initiation Protocol (SIP) is a signalling protocol used for establishing sessions in an IP network. A session could be a simple two-way telephone call or it could be a collaborative multi-media conference session. Over the last couple of years, the Voice over IP community has adopted SIP as its protocol of choice for signaling. SIP is an RFC standard (RFC 3261) from the Internet Engineering Task Force (IETF), the body responsible for administering and developing the mechanisms that comprise the Internet. The IETF's philosophy is one of simplicity: specify only what you need to specify. SIP is very much of this mould; it just initiates, terminates and modifies sessions. This simplicity means that SIP scales, it is extensible, and it sits comfortably in different architectures and deployment scenarios.
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- What Are Some disadvantages of VoIP?
- If you're considering replacing your traditional telephone service with VoIP, there are some possible differences:
- Some VoIP services don't work during power outages and the service provider may not offer backup power.
- Not all VoIP services connect directly to emergency services through 9-1-1. For additional information, see www.voip911.gov.
- VoIP providers may or may not offer directory assistance/white page listings.
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- Whats AGI?
- AGI stands for Asterisk Gateway Interface. AGI allows Asterisk to launch external programs written in any language to control a telephony channel, play audio, read DTMF digits, etc. by communicating with the AGI protocol on stdin and stdout.
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- VoIP tends to be relatively inexpensive. Why?
- VoIP calls are just another application riding over the Internet. And these calls are unregulated. So at their core, they are no different from e-mails, instant messages or Web pages, which all can be distributed for free between Internet-connected machines. Those include computers and wireless devices, such as cell phones and handhelds, that are set up to receive online information.
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- Why do some VoIP services cost money, and why are some free?
- A VoIP service can connect users not only with other VoIP customers but also with phone services that are offline, such as those that use traditional landline networks and wireless cell phone networks. For those calls, VoIP service providers must pay access fees to the landline and wireless operators. Those charges are passed along to VoIP customers. VoIP services that stay on the Internet--calls that are between personal computers with VoIP service--are free.
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- What different types of CODECS are there?
- A Codec converts an analog signal to a digital one for transmission over a data network. The following Codecs are in use today.
- GSM - 13 Kbps (full rate), 20ms frame size
- iLBC - 15Kbps,20ms frame size: 13.3 Kbps, 30ms frame size
- ITU G.711 - 64 Kbps, sample-based. Also known as alaw/ulaw
- ITU G.722 - 48/56/64 Kbps
- ITU G.723.1 - 5.3/6.3 Kbps, 30ms frame size
- ITU G.726 - 16/24/32/40 Kbps
- ITU G.728 - 16 Kbps
- ITU G.729 - 8 Kbps, 10ms frame size
- Speex - 2.15 to 44.2 Kbps
- LPC10 - 2.5 Kbps
- DoD CELP - 4.8 Kbps
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- What is a STUN Server?
- A STUN (Simple Traversal of User Datagram Protocol [UDP] Through Network Address Translators [NATs]) server allows NAT clients (i.e. computers behind a firewall) to setup phone calls to a VOIP provider hosted outside of the local network.
The STUN server allows clients to find out their public address, the type of NAT they are behind and the internet side port associated by the NAT with a particular local port. This information is used to set up UDP communication between the client and the VOIP provider and so establish a call. The STUN protocol is defined in RFC 3489.
The STUN server is contacted on UDP port 3478, however the server will hint clients to perform tests on alternate IP and port number too (STUN servers have two IP addresses). The RFC states that this port and IP are arbitrary
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- What are the benefits of an IP PBX?
- Much easier to install & configure than a proprietary phone system
- Easier to manage because of web based configuration interface
- No need for separate phone wiring
- Allows users to hot plug their phone anywhere in the office - users simply take their phone, plug it into the nearest ethernet port and keep their existing number!
- Allows easy roaming - calls can be diverted anywhere in the world because of the SIP protocol characteristics
- Significant cost reduction by leveraging Internet
- SIP standard eliminates proprietary, expensive phones
- Scalable
- Better reporting
- Better overview of system status and calls
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- What is a SIP server?
- A SIP server is the main component of an IP PBX, dealing with the setup of all SIP calls in the network. A SIP server is also referred to as a SIP Proxy or a Registrar..
Can you list all known sip responses?
1xx = informational responses
- 100 Trying
- 180 Ringing
- 181 Call Is Being Forwarded
- 182 Queued
- 183 Session Progress
2xx = success responses
- 200 OK
- 202 accepted: Used for referrals
3xx = redirection responses
- 300 Multiple Choices
- 301 Moved Permanently
- 302 Moved Temporarily
- 305 Use Proxy
- 380 Alternative Service
4xx = request failures
- 400 Bad Request
- 401 Unauthorized: Used only by registrars. Proxys should use proxy authorization 407
- 402 Payment Required (Reserved for future use)
- 403 Forbidden
- 404 Not Found: User not found
- 405 Method Not Allowed
- 406 Not Acceptable
- 407 Proxy Authentication Required
- 408 Request Timeout: Couldn't find the user in time
- 410 Gone: The user existed once, but is not available here any more.
- 413 Request Entity Too Large
- 414 Request-URI Too Long
- 415 Unsupported Media Type
- 416 Unsupported URI Scheme
- 420 Bad Extension: Bad SIP Protocol Extension used, not understood by the server
- 421 Extension Required
- 423 Interval Too Brief
- 480 Temporarily Unavailable
- 481 Call/Transaction Does Not Exist
- 482 Loop Detected
- 483 Too Many Hops
- 484 Address Incomplete
- 485 Ambiguous
- 486 Busy Here
- 487 Request Terminated
- 488 Not Acceptable Here
- 491 Request Pending
- 493 Undecipherable: Could not decrypt S/MIME body part
5xx = server errors
- 500 Server Internal Error
- 501 Not Implemented: The SIP request method is not implemented here
- 502 Bad Gateway
- 503 Service Unavailable
- 504 Server Time-out
- 505 Version Not Supported: The server does not support this version of the SIP protocol
- 513 Message Too Large
6xx = global failures
- 600 Busy Everywhere
- 603 Decline
- 604 Does Not Exist Anywhere
- 606 Not Acceptable
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- What is a SIP-URI?
- SIP URI = sip:x@y:Port
Where x=Username and y=host (domain or IP)
Examples:
sip:voip.tech@192.168.1.18
sip:sales@nextstag.com
sip:2403223560@nextstag.com
The SIP URI standard has been defined in the RFC 3261 standard.
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