- [Instructor] Voice over internet protocol or VoIP
is a method to carry voice traffic digitally over a network.
Initially, providers released services
that emulated traditional phone systems,
changing out analog lines for digital.
More recently, closed services like Skype
have started offering new alternatives.
Session Initiation Protocol is a signaling
and control protocol for communications.
It is most commonly used for voice and video delivery
over an IP network, but can also carry instant messaging.
SIP is primarily responsible for establishing
and terminating these communications,
but not actually carrying voice traffic.
SIP uses TCP and UDP port 5060 for plaintext traffic
and uses TCP and UDP port 5061 for TLS encrypted traffic.
SIP packets employ a structure very similar to HTTP,
including header fields and encoding rules.
SIP resources are marked with a uniform resource identifier,
much like web resources.
The URI scheme employed is usually in the form of
sip:username:password@host:port.
SIP will often be employed on a digital office phone,
also known as a user agent,
to terminate a local or remote SIP proxy server.
A proxy acts both as a client and a server
taking a call request
and forwarding it to the nearest endpoint it knows.
Not only will the call control
between the handset and proxy server utilize SIP,
but often various gateways will also.
A gateway is a method to connect to other mediums
like outside providers
or the public switch telephone network or PSTN.
H.232 is also a signal and control protocol
for communication services.
It commonly uses TCP and UDP ports from 1718 to 1720.
H.232 was originally developed
to facilitate teleconferencing,
but was quickly adjusted to be more encompassing.
H.323 is most often used today
as a gateway for VoIP services.
The Media Gateway Control Protocol is a plaintext signaling
and control protocol.
As the name implies, it acts as a gateway
allowing a phone system to interface with a PSTN.
Unlike H.323 gateways,
MGCP gateways have very little intelligence
as they act more as a conduit
for the phone system to control.
It utilizes TCP port 2428 and UDP port 2427.
The Real-Time Transfer Protocol
is what actually transfers voice and video traffic.
RTP works in conjunction with the above signaling protocols
as well as the RTP Control Protocol.
RTCP periodically sends statistical information
to participants in a media stream.
It carries no voice traffic.
It was originally developed in 1996,
but was revised in 2003 as RFC 3550.
RTP can detect out-of-sequence packets
and can perform jitter correction.
It also allows for multicast delivery
to multiple destinations simultaneously.
RTP employs an error concealment algorithm,
which means it can sustain some packet loss
without users noticing.
Couple this with a need for audio packets
to be transmitted as quickly as possible
and it's clear why RTP employs UDP
as its delivery mechanism.
RTP has a broad range of ports assigned
from 16387 to 32767 UDP.
An RTP stream should be an even port number
and the next odd port should be the RTCP port.
The RTP profile type specifies the codecs
used to encode the audio
for the payload portion of the RTP packet.
This can be methods such as G.711, G.723, G.726 and so on.
There are a plethora of hosted SIP services
out on the web now.
For smaller organizations, nothing more than phones
and an internet connection are necessary.
For larger organizations,
a single server can terminate SIP phones
and also act as a voicemail server.
Larger enterprises generally utilize multiple SIP
and voicemail servers.
Having VoIP experience is often a plus on a resume
so don't hesitate to get your hands on some.