Training:How does VOIP Work - Revision history

Tyler: /* Network Address Translation – NAT */

2023-06-12T19:12:03Z

Network Address Translation – NAT

Tyler: /* VoIP Protocols */

2023-06-12T18:59:37Z

VoIP Protocols

← Older revision		Revision as of 18:59, 12 June 2023
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	= VoIP Protocols =		= VoIP Protocols =

	~~There are a number~~ of ~~protocols that may be employed in order to provide for~~ VoIP communication ~~services~~. ~~In this section~~, we ~~will focus on~~ IPitomy SIP Trunking ~~since it is~~ the protocol ~~of choice~~ for all IPitomy devices ~~now being deployed~~ as well as third party ~~manufacturers in the~~ industry.		The success of VoIP communication hinges on employing an appropriate set of protocols. Here, we'll discuss IPitomy SIP Trunking, the preferred protocol for all IPitomy devices currently in circulation as well as many other third-party industry offerings.

	~~Virtually every device in the world uses a standard called~~ Real-Time Protocol (RTP) for transmitting audio and video packets between ~~communicating~~ computers. RTP ~~is defined~~ by ~~the~~ open standards ~~that are set using~~ various ~~standards~~ documents~~. RTP also addresses~~ issues like packet order and ~~provides~~ mechanisms ~~(via~~ the Real-Time Control Protocol, to ~~help~~ address delay and jitter.		The Real-Time Protocol (RTP) is a standard used globally by nearly every device for transmitting audio and video packets between computers. RTP, guided by open standards outlined in various documents, manages issues like packet order and employs mechanisms such as the Real-Time Control Protocol to address delay and jitter.

	Before ~~audio or video~~ media can flow between two devices, ~~various~~ protocols ~~must be employed~~ to ~~find~~ the remote device and to negotiate the ~~means by which~~ media ~~will~~ flow ~~between the two devices~~. ~~The~~ protocols ~~that are central to this process~~ are ~~referred to~~ as call-signaling protocols, the ~~most popular of which is~~ Session Initiation Protocol (SIP).		Before media can flow between two devices, protocols are utilized to locate the remote device and negotiate the media flow methods. These crucial protocols are known as call-signaling protocols, with the Session Initiation Protocol (SIP) being the most widely used.

	~~Advantage of IPitomy SIP Trunks~~

	*SIP is a simple protocol (at least for machines).  It is a text base protocol that is designed to be easily read and lends itself well to troubleshooting by being able to see what is in the packets without having to completely decompile the software,		Advantages of IPitomy SIP Trunks
	*SIP ~~works very similar to email; The addressing is very similar.~~
	*SIP calls are pure digital voice.  In its native world, there is no distortion, no delay and no echo.
	*Instant Scalability; Since IPitomy uses an IP PBX platform we can increase or decrease capacity with a simple, easy process in the admin GUI
	*Many different devices can interoperate in a network providing a wide variety of choice for users that extends far beyond what any single vendor can provide. International dialing.
	*Location of users is irrelevant; easily deploy remote users anywhere in the world without losing direct connectivity. The only requirement is an adequate internet connection.

	~~Anatomy~~ of a ~~SIP Call~~		SIP, a text-based protocol, is relatively straightforward for machines to understand. Its easy readability facilitates troubleshooting by allowing the inspection of packet contents without needing to decompile the software entirely. SIP's operation mirrors that of email, making it familiar and intuitive. Its addressing methods are particularly similar. SIP calls offer undiluted digital voice quality, free of distortion, delay, or echo in its native environment. Instant Scalability: Given IPitomy's IP PBX platform, capacity can be easily adjusted via a simple process in the admin GUI. Interoperability with a wide range of devices provides users with choices far beyond what any single vendor can offer. International dialing is supported. User location is inconsequential; remote users can be deployed globally without losing direct connectivity. An adequate internet connection is the only requirement.

	A SIP Call has a signaling component and a Voice component.  The signaling path is different from the actual voice transmission path.  The signaling and voice transmission are unique for each call.  All call setup and teardown signaling is over port 5060.  All voice transmissions are on a range of ports 10,000 to 20,000.  The ports are virtual ports and are simply part of the protocol for communicating over TCPIP.

	~~Setting the Stage~~		Understanding a SIP Call

	Before a SIP call can be placed, there needs to be SIP endpoints that have the ability to” find” and “be found by” other SIP endpoints in order to make and receive calls.  For the purposes of this training guide, we will limit our SIP ~~examples to endpoints that register to an IPitomy IP PBX.  While it is possible to~~ call ~~from one SIP endpoint to another directly using~~ a ~~peer-to-peer method, most calls are facilitated through an IP PBX or soft switch to make dialing simple~~ and ~~easy just like~~ a ~~PSTN call~~.~~ ~~ The ~~big advantage of having an IP PBX is that the user does not have to dial an IP address to~~ call ~~another endpoint~~ and ~~all of the call information can be stored~~ for ~~reporting etc. The IP PBX will handle routing~~ all ~~of the~~ calls ~~to their destination on~~ the ~~PSTN or~~ to ~~local and remote extensions~~.~~  Users simply dial phone numbers and extension numbers just like a legacy PBX system. Although not recommended due~~ to ~~their EOL an IPitomy IP PBX supports analog PSTN lines and T1/PRI cards~~.~~ ~~		A SIP call comprises a signaling component and a Voice component. The paths for signaling and actual voice transmission differ for each call. Setup and teardown signaling for all calls operate over port 5060. Voice transmissions are conducted within the range of ports 10,000 to 20,000. These are virtual ports integral to TCPIP communication.

	In order to be part of the PBX database, SIP endpoints register with the PBX.  Once they are registered, they have the ability to dial phone numbers and be called by other endpoints.  The registration can be from phones on the Local Area Network (LAN) or from anywhere on the Internet.

	~~Now that the phones are registered with the PBX,~~ a call can be initiated or received.  To start the call, the endpoint sends an invite to the server to ask the other endpoint if it is available to take a call.  This takes place on the signaling port (port 5060).  If the other endpoint is ready to accept the call, it sends an acknowledgement back to the initiating phone.  The initiating phone then sends the call information telling the other phone which ports to commence the RTP (voice) session on.  The RTP session is opened using the ports communicated by the initiating call.		Preconditions for a SIP Call

	~~When the~~ call ~~is over~~, ~~one~~ of ~~the~~ endpoints ~~sends~~ a ~~bye message~~ and ~~the~~ call ~~hangs up~~.~~  That is~~ a ~~pretty simple description of how~~ SIP ~~makes a call.  When~~ the ~~call is on the LAN it does not have~~ to ~~go through~~ the ~~router~~.~~  The PBX will handle telling the~~ endpoints ~~which ports to use to connect~~ the ~~RTP~~ (~~voice~~) ~~stream~~.~~ ~~		Before a SIP call can occur, SIP endpoints capable of finding and being found by other SIP endpoints are needed. We'll restrict our SIP examples to endpoints that register to an IPitomy IP PBX for this training guide. While peer-to-peer SIP calls are possible, most are facilitated via an IP PBX or soft switch for a PSTN-like ease of dialing. The advantage of an IP PBX is that it simplifies calling another endpoint, allows call information to be stored for reporting, and manages call routing to local and remote extensions. Users dial phone numbers and extension numbers as they would with a legacy PBX system. Although no longer recommended due to their End-Of-Life status, an IPitomy IP PBX still supports analog PSTN lines and T1/PRI cards. SIP endpoints must register with the PBX to be included in the PBX database. Once registered, they can dial phone numbers and receive calls from other endpoints. These endpoints can be phones on the Local Area Network (LAN) or any location on the internet.

	~~When~~ the ~~call~~ is to ~~a remote~~ phone, the ~~PBX knows~~ the phone is ~~outside~~ of the ~~firewall~~.~~ ~~ This is ~~when~~ the router ~~needs~~ to ~~have ports configured~~ for signaling and RTP traffic.~~ ~~
			Starting a SIP Call

			To start a call, the endpoint sends an invite to the server requesting the other endpoint's availability. This occurs on the signaling port (port 5060). If the other endpoint is ready, it sends an acknowledgement back to the initiating phone, which then sends the call information instructing the other phone on the ports to commence the RTP (voice) session. The RTP session is opened using the communicated ports.

			Call termination is initiated by one of the endpoints sending a "bye" message, causing the call to hang up. This is a simplified explanation of a SIP call's lifecycle. When the call occurs on the LAN, it bypasses the router. The PBX instructs the endpoints on the ports to connect the RTP (voice) stream.

			For calls to remote phones, the PBX understands that the phone is beyond the firewall. Router port configuration becomes necessary for signaling and RTP traffic at this stage.

	Signaling Port 5060		Signaling Port 5060

	~~When the ports are properly configured,~~ port 5060 is forwarded in the router to the PBX ~~systems~~ IP address ~~on the LAN.  This allows~~ the PBX to send signals to ~~the~~ remote phones ~~as well as~~ receive requests from them.		Proper port configuration enables port 5060 to be forwarded in the router to the PBX system's LAN IP address, allowing the PBX to send signals to remote phones and receive requests from them.

	RTP Ports 10,000 – 20,000 Port Range Forwarding		RTP Ports 10,000 – 20,000 Port Range Forwarding

	Once ~~the~~ call is setup ~~using the~~ signaling on Port 5060, ~~the~~ RTP is ~~setup~~ using a range of ports ~~that are~~ forwarded to the PBX LAN IP address.~~  Using~~ Port Range Forwarding in the Router, the range of Ports 10,000 – 20,000 ~~is forwarded~~ for this purpose.		Once call setup occurs via signaling on Port 5060, RTP is set up using a range of ports forwarded to the PBX LAN IP address. The Port Range Forwarding feature in the Router is used to forward the range of Ports 10,000 – 20,000 for this purpose.

	Each call requires two ports for RTP; one for sending and one for receiving.~~  SIP sets this up from~~ the initiating phone. The ports ~~in the router~~ are open from ~~the~~ inside of the firewall.~~ ~~ The phone ~~on the far end~~ receives the information ~~on what~~ ports to use in the SIP packets.		Each call requires two ports for RTP - one for sending and one for receiving. This is organized by the initiating phone. The router ports are open from inside the firewall. The remote phone receives the information about which ports to use from the SIP packets.

	'''Local Phone Diagram'''		'''Local Phone Diagram'''

Tyler at 18:49, 12 June 2023

2023-06-12T18:49:34Z

← Older revision		Revision as of 18:49, 12 June 2023
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	= How Does VoIP Work?<br/> =		= How Does VoIP Work?<br/> =

	~~The purpose of this~~ section ~~will be to provide a very high-level~~ overview of Voice over IP (VoIP).		This section provides an introductory overview of Voice over Internet Protocol (VoIP), a technology that enables the transmission of voice communications via IP networks.

	~~ Many people have used~~ a computer ~~and~~ a ~~microphone to record~~ a human voice ~~or other sounds. The process involves sampling the sound that is heard by the computer~~ at a ~~very~~ high rate (at least 8,000 times per second ~~or more~~) and ~~storing those "~~samples~~" in memory or in a file on the computer~~. ~~Each sample of sound is just a very tiny bit of the person's voice or other sound recorded by the computer. The computer has the ability to take all of those~~ samples ~~and play them~~, ~~so that the listener can hear what was recorded~~.		VoIP works on the principle of audio sampling, where a computer records a sound (such as a human voice) at a high rate (typically at least 8,000 times per second) and converts these audio samples into digital data. Unlike traditional recording, where these samples are stored locally, VoIP sends these samples over an IP network to be played back on a different device.

	VoIP ~~is based on the same idea~~, ~~but~~ the ~~difference is that~~ the ~~audio~~ samples are ~~not stored locally. Instead~~, ~~they are sent over~~ the ~~IP network~~ to ~~another device and played there~~.		The process of making VoIP function efficiently involves several key steps. Initially, the computer compresses the recorded sound samples to minimize the space they require, focusing particularly on voice frequencies. This compression and decompression process is handled by a tool known as a CODEC (compressor/decompressor). Numerous CODECs are available, and VoIP uses those optimized for voice compression, significantly reducing the bandwidth required compared to uncompressed audio.

	~~Of course~~, ~~there is much more required in order to make VoIP work. When recording~~ the ~~sound~~ samples, ~~the computer might compress those sounds so that they require less space and will certainly record only~~ a ~~limited frequency range~~. ~~There are a number~~ of ~~ways to compress~~ audio, ~~the algorithm for which is referred to as a "compressor/de-compressor",~~ or ~~simply CODEC. Many CODECs exist for a variety of applications (e.g., movies and sound recordings) and, for VoIP,~~ the CODECs are optimized for compressing voice, which significantly reduce the bandwidth used compared to an uncompressed audio stream. Speech CODECs are optimized to improve spoken words in the frequency range of human speech.		After compression, the samples are grouped into larger units and inserted into data packets ready for transmission over the IP network, a process known as packetization. A typical IP packet can contain 10 or more milliseconds of audio, with 20 or 30 milliseconds being the most common.

	~~Once the sound is recorded by the computer and compressed into very small samples~~, ~~the samples are collected together into larger chunks~~ and ~~placed into data packets for transmission over~~ the ~~IP network. This process is referred to packetization. Generally~~, a ~~single IP packet will contain 10 or more milliseconds of audio, with 20~~ or ~~30 milliseconds being most common~~.		A comparison could be made to sending postcards through traditional mail. Each postcard (packet) carries a limited amount of information. Sending a lengthy message would require multiple postcards (packets), and to ensure they can be assembled correctly at the destination, they are organized using a sequence number or similar mechanism.

	A good comparison is to think of a packet as a postcard sent via postal mail. A postcard contains just a limited amount of information. To deliver a very long message, one must send a lot of postcards. Of course, the post office might lose one or more postcards. One also has to assemble the received postcards in order, so some kind of mechanism must be used to properly organize the postcards, such as placing a sequence number on the bottom right corner. One can think of data packets in an IP network as postcards.

	[[File:How_voip_1.gif\|alt=]]		[[File:How_voip_1.gif\|alt=]]

Mike Lunn: /* VoIP Protocols */

2022-06-13T15:32:02Z

VoIP Protocols

← Older revision		Revision as of 15:32, 13 June 2022
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	'''Remote Phone Diagram'''		'''Remote Phone Diagram'''

	[[File:Academy - RDP v3.gif\|alt=]]		[[File:Academy - RDP v5.gif\|alt=]]

	== Network Address Translation – NAT ==		== Network Address Translation – NAT ==

Mike Lunn: /* VoIP Protocols */

2022-06-13T14:33:43Z

VoIP Protocols

← Older revision		Revision as of 14:33, 13 June 2022
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	'''Remote Phone Diagram'''		'''Remote Phone Diagram'''

	[[File:Academy - RDP.gif\|alt=]]		[[File:Academy - RDP v3.gif\|alt=]]

	== Network Address Translation – NAT ==		== Network Address Translation – NAT ==

Mike Lunn: /* VoIP Protocols */

2022-06-13T14:25:28Z

VoIP Protocols

← Older revision		Revision as of 14:25, 13 June 2022
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	'''Remote Phone Diagram'''		'''Remote Phone Diagram'''

	[[File:Academy - RDP.gif\|alt=]]		[[File:Academy - RDP.gif\|alt=]]

Mike Lunn: /* VoIP Protocols */

2022-06-13T14:24:46Z

VoIP Protocols

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	[[File:5060.png\|alt=]]		[[File:5060.png\|alt=]]

			'''Remote Phone Diagram'''
	[[File:Academy - RDP.gif\|alt=]]		[[File:Academy - RDP.gif\|alt=]]

Mike Lunn: /* VoIP Protocols */

2022-06-13T14:23:56Z

VoIP Protocols

← Older revision		Revision as of 14:23, 13 June 2022
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	*SIP works very similar to email; The addressing is very similar.		*SIP works very similar to email; The addressing is very similar.
	*SIP calls are pure digital voice.  In its native world, there is no distortion, no delay and no echo.		*SIP calls are pure digital voice.  In its native world, there is no distortion, no delay and no echo.
	*Instant Scalability; Since IPitomy uses an IP-PBX we can increase or decrease capacity with a simple, easy process in the admin GUI		*Instant Scalability; Since IPitomy uses an IP PBX platform we can increase or decrease capacity with a simple, easy process in the admin GUI
	*Many different devices can interoperate in a network providing a wide variety of choice for users that extends far beyond what any single vendor can provide. International		*Many different devices can interoperate in a network providing a wide variety of choice for users that extends far beyond what any single vendor can provide. International dialing.
	*Location of users is irrelevant.~~  As long as they have access to broadband~~.		*Location of users is irrelevant; easily deploy remote users anywhere in the world without losing direct connectivity. The only requirement is an adequate internet connection.

	Anatomy of a SIP Call		Anatomy of a SIP Call
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	Setting the Stage		Setting the Stage

	Before a SIP call can be placed, there needs to be SIP endpoints that have the ability to” find” and “be found by” other SIP endpoints in order to make and receive calls.  For the purposes of this training guide, we will limit our SIP examples to endpoints that register to an IPitomy IP PBX.  While it is possible to call from one SIP endpoint to another directly using a peer-to-peer method, most calls are facilitated through an IP PBX or soft switch to make dialing simple and easy just like a PSTN call.  The big advantage of having an IP PBX is that the user does not have to dial an IP address to call another endpoint and all of the call information can be stored for reporting etc.  The IP PBX will handle routing all of the calls to their destination on the PSTN or to local and remote extensions.  Users simply dial phone numbers and extension numbers just like a legacy PBX system.~~  An~~ IP PBX supports analog PSTN lines, T1/PRI ~~lines, DID’s and SIP Trunks~~.		Before a SIP call can be placed, there needs to be SIP endpoints that have the ability to” find” and “be found by” other SIP endpoints in order to make and receive calls.  For the purposes of this training guide, we will limit our SIP examples to endpoints that register to an IPitomy IP PBX.  While it is possible to call from one SIP endpoint to another directly using a peer-to-peer method, most calls are facilitated through an IP PBX or soft switch to make dialing simple and easy just like a PSTN call.  The big advantage of having an IP PBX is that the user does not have to dial an IP address to call another endpoint and all of the call information can be stored for reporting etc. The IP PBX will handle routing all of the calls to their destination on the PSTN or to local and remote extensions.  Users simply dial phone numbers and extension numbers just like a legacy PBX system. Although not recommended due to their EOL an IPitomy IP PBX supports analog PSTN lines and T1/PRI cards.

	In order to be part of the PBX database, SIP endpoints register with the PBX.  Once they are registered, they have the ability to dial phone numbers and be called by other endpoints.  The registration can be from phones on the Local Area Network (LAN) or from anywhere on the Internet.		In order to be part of the PBX database, SIP endpoints register with the PBX.  Once they are registered, they have the ability to dial phone numbers and be called by other endpoints.  The registration can be from phones on the Local Area Network (LAN) or from anywhere on the Internet.
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			[[File:Academy - RDP.gif\|alt=]]
	[[File:~~HowVOIP6~~.gif\|~~none\|HowVOIP6.gif~~]]

	== Network Address Translation – NAT ==		== Network Address Translation – NAT ==

Mike Lunn: /* VoIP Protocols */

2022-06-13T13:19:52Z

VoIP Protocols

← Older revision		Revision as of 13:19, 13 June 2022
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	= VoIP Protocols =		= VoIP Protocols =

	There are a number of protocols that may be employed in order to provide for VoIP communication services. In this section, we will focus on SIP since it is the protocol of choice for ~~most~~ devices now being deployed in the industry.		There are a number of protocols that may be employed in order to provide for VoIP communication services. In this section, we will focus on IPitomy SIP Trunking since it is the protocol of choice for all IPitomy devices now being deployed as well as third party manufacturers in the industry.

	Virtually every device in the world uses a standard called Real-Time Protocol (RTP) for transmitting audio and video packets between communicating computers. RTP is defined by the open standards that are set using various standards documents. RTP also addresses issues like packet order and provides mechanisms (via the Real-Time Control Protocol, to help address delay and jitter.		Virtually every device in the world uses a standard called Real-Time Protocol (RTP) for transmitting audio and video packets between communicating computers. RTP is defined by the open standards that are set using various standards documents. RTP also addresses issues like packet order and provides mechanisms (via the Real-Time Control Protocol, to help address delay and jitter.
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	Before audio or video media can flow between two devices, various protocols must be employed to find the remote device and to negotiate the means by which media will flow between the two devices. The protocols that are central to this process are referred to as call-signaling protocols, the most popular of which is Session Initiation Protocol (SIP).		Before audio or video media can flow between two devices, various protocols must be employed to find the remote device and to negotiate the means by which media will flow between the two devices. The protocols that are central to this process are referred to as call-signaling protocols, the most popular of which is Session Initiation Protocol (SIP).

	Advantage of SIP		Advantage of IPitomy SIP Trunks

	*SIP is a simple protocol (at least for machines).  It is a text base protocol that is designed to be easily read and lends itself well to troubleshooting by being able to see what is in the packets without having to completely decompile the software,		*SIP is a simple protocol (at least for machines).  It is a text base protocol that is designed to be easily read and lends itself well to troubleshooting by being able to see what is in the packets without having to completely decompile the software,
	*SIP works very similar to email~~.&nbsp~~; The addressing is very similar.		*SIP works very similar to email; The addressing is very similar.
	*SIP calls are pure digital voice.  In its native world, there is no distortion, no delay and no echo.		*SIP calls are pure digital voice.  In its native world, there is no distortion, no delay and no echo.
	*Many different devices can interoperate in a network providing a wide variety of choice for users that extends far beyond what any single vendor can provide.		*Instant Scalability; Since IPitomy uses an IP-PBX we can increase or decrease capacity with a simple, easy process in the admin GUI
			*Many different devices can interoperate in a network providing a wide variety of choice for users that extends far beyond what any single vendor can provide. International
	*Location of users is irrelevant.  As long as they have access to broadband.		*Location of users is irrelevant.  As long as they have access to broadband.

Mike Lunn: /* Jitter in Packet Voice Networks */

2022-05-23T14:19:10Z

Jitter in Packet Voice Networks

← Older revision		Revision as of 14:19, 23 May 2022
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	If the jitter is so large that it causes packets to be received out of the range of this buffer, the out-of-range packets are discarded and dropouts are heard in the audio. For losses as small as one packet, the DSP interpolates what it thinks the audio should be and no problem is audible. When jitter exceeds what the DSP can do to make up for the missing packets, audio problems are heard.		If the jitter is so large that it causes packets to be received out of the range of this buffer, the out-of-range packets are discarded and dropouts are heard in the audio. For losses as small as one packet, the DSP interpolates what it thinks the audio should be and no problem is audible. When jitter exceeds what the DSP can do to make up for the missing packets, audio problems are heard.

	This diagram illustrates how excessive jitter is handled.[[File:~~HowVOIP4.gif\|~~HowVOIP4.gif\|alt=\|none\|frame]]		This diagram illustrates how excessive jitter is handled.
			[[File:HowVOIP4.gif\|alt=\|none\|frame]]