1. VOIP – AN INSIGHT INTO A PROGRESSING TECHNOLOGY
David Tarrant
Researcher in Intelligence, Agents and Multimedia
School of Electronics and Computer Science
University of Southampton, SO17 1BJ.
dt302@ecs.soton.ac.uk
www.ecs.soton.ac.uk/~dt302
ABSTRACT In this report we address the many developing
technologies behind VoIP and which ones are now
Technology behind the Voice over Internet Protocol leading the field into the next era of telecoms
(VoIP) has been accelerating rapidly over the past communications. Beginning with a quick look at the
few years however is still not ready for full older circuit switching networks such as the
deployment. VoIP provides a Peer 2 Peer (P2P) link worldwide PSTN and how VoIP has been developed
between users for making voice calls over the based on this technology and what improvements
internet and servers are beginning to appear in all have been made. An outline of the technology
areas of the community, providing different levels of behind VoIP and different uses is then drawn before
service to users. However these services can be looking at the supporting server and client
based on different protocols and standards making applications. We look at problems that have arisen
interoperability between them difficult. This paper during the construction of VoIP and outline what
provides an overview on the different standards and actions are being taken to resolve these. A case
protocols used in VoIP asking: Where will these study from the research currently taking place at the
take us in the future? What advantages does VoIP University of Southampton looks at some of the
have to offer over current telephony? And what issues that have arisen and shows where
implications are there for existing telecoms commercial interest is being focused. This will
companies? enable a strong outlook to be drawn on where the
future of the VoIP technology lies and how users
Keywords can get involved.
VoIP, SIP, H.323, Telephony.
2. BRIEF HISTORY OF TELEPHONY
1. INTRODUCTION COMMUNICATIONS
Voice over IP (VoIP), otherwise known as IP Traditional telephone networks are connected via
telephony, is the delivery of voice information over copper core to a circuit switched network. When a
packet switched networks. This means sending user dials a number this causes a route to be set up
voice information in digital form in discrete packets based on that unique number to another phone
rather than in the traditional circuit-committed somewhere in the world. This global public switch
protocols of the public switched telephone network telephone network (PSTN) is divided into individual
(PSTN). A major advantage of VoIP is that it can local switch areas, which combined together make
avoid the tolls charged by ordinary telephone up a Local Access and Transport Area (LATA).
service by utilising fixed charge IP network services Each country can be dived into many geographic
such as broadband. Recent development with LATAs that are controlled by local exchange carriers
technologies such as SIP and hardware supporting (LEC). Depending on origin and destination, a call
this standard has resulted in the production of a may traverse many switches and across multiple
number of commercially marketed SIP handsets, LATAs to be established. Once established a
both for wired and wireless networks, removing the
need for a PC or laptop running a software handset, National Call
or “softphone”. A subscription to a local server from
a SIP handset or softphone provides you with all the
normal telephony features including voice and fax,
0 2 380 595656
as well as opportunities for text and video services.
LEC Type Geographic LATA
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provided that copies are not made or distributed for profit or
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th LEC Type
4 Annual Multimedia Systems Conference, Electronics and
Geographic LATA
Computer Science, University of Southampton
(Southampton)
© 2004 Electronics and Computer Science, University of
Southampton Figure 1 – Breakdown of UK phone number.

2. singular connection is thus achieved between the congested and quickest route. Figure 2 outlines this
two clients via this route and voice packets can now process and shows two clients connecting over the
be transmitted. Figure 1 shows a break down of internet.
phone number into its many parts showing how a
call would be established between 2 clients. Call CLIENT DISCOVERY
costs can be implied by simply finding out which VoIP SERVERS
LATAs have been traversed and applying charges
appropriately [5].
INTERNET
2.1. Telephony into VoIP
VoIP is based heavily on the already existent
structure of the worldwide PSTN, however the
active environment is the internet and thus VoIP has VoIP CALL – DIRECT CONNECTION
been tuned to use existing network protocols where
CLIENT 1 CLIENT 2
available. Like the PSTN network a user will be
connected to a local exchange (server) which in turn Figure 2 – VoIP Call example (discovery and connection)
is connected to other servers around the world.
This decentralized architecture is ideal for end to
These servers are able to communicate freely with
end connections of only two users, however
each other in order to find and connect users [5].
connection and management of conference calls
VoIP has two main deployment methods based
becomes more of a challenge. Making multipoint
upon protocols from different developers. The ITU-T
calls involves using IP multicast to transmit data to
recommendation H.323 [3] follows a client server
many users, which means that users must be able
architecture much like the worldwide PSTN. Clients
to transmit and receive multicast packets at their
interact both for data transport and control with a
location.
small number of servers which coordinate and
control the session. The IETF recommends the
Session Initiation Protocol (SIP) [4] which is a highly
decentralized architecture where servers are only
used to locate users. A peer to peer link over the
internet can then be established between the users
without the need for an expensive powerful server.
3. THE SESSION INITIATION PROTOCOL
(SIP)
SIP (Session initiation protocol) is an Internet
standard specified by the Internet Engineering Task
Force (IETF) in RFC 2543 [4]. SIP is used to initiate,
manage, and terminate interactive sessions
between one or more users on the Internet. SIP
borrows heavily from HTTP and the e-mail protocol Figure 3 – Structure of SIP
SMTP, providing scalability, extensibility, flexibility,
and capabilities for creation of new services. As a While servers are required to carry out some of the
result SIP is increasingly used for Internet telephony more complex SIP features such as transcoding, it
signalling, in gateways, PC phones, softswitches, is possible set up point to point or multicast
and softphones, however is not limited to Internet conference calls without the need for a server. SIP
telephony and can be used to initiate and manage has been designed specifically to allow clients to
any type of session, including video, interactive make use of IP packets for both control and data
games, and text chat. SIP takes advantage of the transport within calls.
underlying technology of the internet, harnessing
A generic SIP call involves a SIP User Agent (UA)
this where possible so as to decentralize any
locating a user on a registrar server (VoIP server)
dependencies on the SIP server. A good example of
and then issuing an invitation to them via a proxy
which is how users are connected over a SIP
server making use of any redirect servers where
network: Unlike PSTN once the two users are
appropriate. A successful SIP invitation consists of
located the call is not connected via the servers or
two messages: INVITE followed by an ACK. The
the route taken in order to find the users. The
INVITE message contains a session description
internet already contains a route optimisation
from the UA containing information on which type of
framework at the packet level and thus users are
media to caller wishes to use and can accept for the
connected direct to each other using a peer to peer
call. Media types, often referred to as codecs
link. By default packets will traverse the least

3. included many such as GSM and the ITU codecs, these being those which establish and negotiate the
some of which are already in use on mobile phone calling properties and media transmission types in
networks and in other commercial voice use. H.450 defines a generic functional protocol on
applications. This capability enables SIP to take full top of H.225 for all supplementary services and
advantage of current technology and be integrated provides the only abstraction layer to H.323 where
where possible. extra services can be harnessed in a call.
Being based heavily on the SMTP and HTTP
protocols, SIP adopts many of the methods and
usability from these. None more so than user
location, as well as each user having a unique
number, users can also have a user@host.domain
address which is simply aliased to their number.
Finally SIP also provides a Session Announcement
Protocol (SAP) and the Session Description
Protocol (SDP) which support the establishment of
multiparty conferencing sessions. SDP defines the
description of multimedia sessions, while SAP
enables periodic multicasting of information about
active sessions. Together these enable third party
users to join an already established session within a
given time frame. Figure 4 – Structure of H.323
Figure 3 depicts the overall IETF SIP protocol suite H.323 can have a number of servers to perform
and the many extensions available, with space left different tasks depending on the scenario. Typically
for many more which are being worked upon by the an H.323 gatekeeper (GK) performs many of tasks
IETF as internet drafts. SIP itself simply provides a equivalent to the SIP proxy server providing address
small number of text based messages to be translation, RAS control, call redirection and
exchanged in separate transactions between the resource management. H.323 can also create
SIP peer entities. The session itself is described at decentralised point to point links to users for use in
two levels. The SIP protocol contains the parties’ calls however the gatekeeper hander the
addresses and protocol processing features (media initialisation and termination of the session rather
types), with the body containing SDP which is a than the individual clients. In decentralised calling
structured, text-based media description format. mode each client must also act as multipoint
Since the message body is transparent to SIP any processor and be able to process media streams,
type of SDP can be transferred, thus not limiting SIP including multicast.
to use in VoIP, but opening it up to use in any
To enable advanced features such as conference
session based application. SIP extensions such as
calling in H.323 further servers must be used to
event notification (RFC 3265), session update (RFC
establish and manage connections between multiple
3311), call transfer and call holding can then be
users. A multipoint controller (MC) establishes an
applied to complete the SIP core framework and
H.245 control connection to each user for
use in VoIP.
negotiation of media communication types. The
4. H.323 multipoint processor (MP) is then able to decode
and retransmit the streams as required. The H.323
H.323 defines system aspect requirements for MC component is responsible for selecting unicast
multimedia communication systems over a packet or multicast media transmission and for choosing
switching network. This includes registration, network/transport addresses.
admission and status (RAS or RTP/RTCP) control,
call setup as defined in H.225.0 and call setup and To establish a call using H.323 the II.323 call
signalling as defined in H.245. H.225.0 defines an signalling procedure has to be carried out to
alias type for carrying any standard Uniform establish valid H.245 connections via the
Resource Locator (URL) [11]. H.323 version 4 [3] gatekeeper. The II.323 call signalling procedure
introduced an H.323-specific URL, which may be begins when an originating H.323 client issues an
used to resolve the address of a network entity to admission request (ARQ) to local gatekeeper in its
which H.323 calls may be directed. Like SIP H.323 domain. When the corresponding confirmation
also supports many audio and video codecs for use message (ACF) is received the call setup procedure
in calls, as well as real time media transport continues with a SETUP and CONNECTION
protocols (RTP and RTCP). message exchange. Upon successful establishment
of a call the clients follow the H.245 capability
Figure 4 depicts the structure of the H.323 protocol exchange procedure to open media channels which
suite showing the compulsory objects in dark tan, both clients are able to support. In later versions (3

4. upwards) of H.323 clients are able to reduce gives a comparison between the many codecs with
signalling overhead by using the Fast Connection bit rate (quality) and bandwidth required for each to
procedure. This “FastStart” procedure is included as be used. With the operation of VoIP on a large scale
an element in the SETUP message sent on call being over the internet a dedicated service for voice
establishment. The “FastStart” element carries the calls is not possible unlike the old PSTN system.
proposed media channel description defining the This can lead to other factors affecting the quality of
media capabilities of the origin of the call allowing service as well as the limitations on ADSL upload
media communication to begin after one round-trip speed. This introduces many problems such as
message exchange instead of three. delay, packet loss, bandwidth limitations and echo.
With a normal ADSL connection (512:256) users are
5. COMPARISON BETWEEN H.323 AND likely to experience latencies of between 80ms and
SIP 400ms on a call, at around 200ms the flow of
conversation becomes distorted. This is mainly due
Both SIP and H.323 protocols can be used to most residential ADSL only supplying minimal
efficiently to connect client to client calls using any upload bandwidth thus limiting the user’s
range of media codecs. By placing a suitable proxy capabilities. All this is likely to change firstly when
server in the middle is it also possible to perform a SDSL becomes more widely available and then as
SIP to H.323 client call. Problems arise when BT themselves switch to IP based networks over the
complex functions such conference calls are next 5 years, known as 21CN [6][7].
attempted. H.323 is based on a centralized server
that uses a set of tightly integrated protocols to 7. VOIP SERVER PACKAGES
control sessions and users connections. In contrast,
SIP is often without a server, and its control There are many different VoIP server packages
mechanisms are much more loosely coupled and available each with their advantages over the other
depend a lot more on the client technology. SIP and each of their own complexity. At the time of
clients are able to join and leave a conference by writing Asterisk1 was the most popular server
using UDP signalling without the need for package providing built-in support for both H.323
centralised control. A central server in SIP is able to and SIP with functionality which can be harnessed
provide easier location of clients as well as after only about 20 minutes for setup. SIP Express
centralised session announcement (SAP) using the Router (SER)2 from iptel.org provides a proxy/router
Session Description Protocol (SDP). SIP provides a for SIP sessions with an optional extensions module
far more abstract protocol than H.323 able to be for the construction of a VoIP server. To set up a
used outside of systems such as VoIP, such as that PSTN like system in SER would take a lot longer
of multicast and unicast video streaming. H.323 is than the 20minutes of Asterisk, however SER
based around only a few recommendations and provides its own scripting language for complete
thus becomes easier to pick up and use for user control of functionality. Finally the last major
developers. A centralized architecture such as that server package is VOCAL3, a much more
of H.323 is more preferable to government services commercial solution with greater support for
which can still keep and eye on usage and enable a business and enterprise users. VOCAL is an open
line to be tapped, this being a requirement of any source project primarily designed as a SIP
ITU phone network. As SIP is not specifically softswitch, however includes translator plug-ins for
designed for telephony it does not have to comply support of H.323 endpoints. The following section
with this ruling as yet [8]. provides a more in depth look at the technology of
the 3 main server technologies and their
6. VOIP CODECS differences.
VoIP codecs are used to convert an analogue voice Being the most popular at the time of writing
signal into a digitally encoded version for Asterisk provides high levels of built in functionality
transmission over the internet. The same codec is which is easy to manage. Its ability to be able to
then used for the opposite purpose at its handle complex dialling plans and a wide range of
destination. Codecs vary greatly in sound quality, voice, fax, text and video codecs for direct
bandwidth required and computational interaction with users is able to seamlessly provide
requirements. Each server, program, gateway, etc switchboard, voicemail and operator support on the
typically supports several different codecs use of server. Asterisk supports both the H.323 and SIP
which use is negotiated upon initialization of the call. standards and most popular codecs, those used for
Server codec support is only required if the server is interaction with the user are shown in Figure 5.
able to interact with the client in operations such as Asterisk also provides the proprietary IAX (Inter-
switchboards and voicemail. Both SIP and H.323 Asterisk eXchange) protocol to enable the
contain abstraction layers supporting a set of 1 Asterisk – http://www.asterisk.org
standard codecs defined by the ITU for voice calls, 2 SER – http://www.iptel.org/ser
being more built into H.323 than SIP. Appendix A 3 VOCAL – http://www.vovida.org/

5. interconnection of multiple Asterisk servers, with the Codec Asterisk SER VOCAL
ability to forward communications between servers G. 711 Y Y
or use one server as backup for another. IAX G.723.1 Y
supplies a facility for VoIP with the same G.726 Y
functionality as an LEC within a PSTN’s LATA. G.729 Y
Asterisk provides a complete worldwide solution GSM 06.10 Y Y
such as that currently provided by the existing
LPC10e Y
PSTN network and is able to seamlessly interface
iLBC Y
with the PSTN network by making use of specifically
Speex Y
designed hardware. By deploying this hardware in
the correct fashion Asterisk is able to fulfil many of Figure 5 –Server support for codecs.
the ITU and FCC regulations i.e. enabling users to
be able to contact the emergency services from any Finally VOCAL from Vovida provides a much more
handset. This is achieved by adding a simple structured enterprise solution which is designed as
extension rule to the system to forward the a SIP softswitch however translators are available to
appropriate numbers onto the correct end users. allow interoperability with H.323 and MGCP
Asterisk provides both a structured number endpoints. The aim of the VOCAL project is to
identification as well as the newer provide a SIP based replacement to the PBX/PSTN
user@host.domain identification which is stored in a without necessarily providing any extra functionality
simple xml type extensions file which Asterisk reads such as that of Asterisk and SER. Vocal is designed
upon startup. Being designed to run on the UNIX to run on a distributed architecture of servers
platform Asterisk contains modules which can be providing redundancy to handle downtime and high
plugged in to enable user, extensions and calling volumes of usage [2]. Not providing any functionality
profiles to be managed by a web interface system such as voicemail at the server end means VOCAL
and harness the UNIX server capabilities. The one does not need any support for codecs, services
downside of the current implementation of Asterisk such as voicemail and call holding are left to the
(version 1.0) is the lack of support for IPv6 which individual clients rather than being supplied
would be required for large scale networks and for centrally. This makes VOCAL the much more
users currently behind a NAT (Network Address extensible system however from a users point a
Translation) or Firewall. view is not the easiest solution. Being an open
source project VOCAL also has to be installed from
SIP Express Router (SER) is a high performance, source as can both Asterisk and SER (which both
configurable, VoIP server supporting only SIP have more binary support however). VOCAL is a
clients and services. SER uses a full scripting very formidable package with the source code
language for its configuration, cutting down on the coming in at 78.1Mb, as opposed to 9.8Mb for SER
number of individual configuration files and or 37Mb for Asterisk.
improving scalability. This comes at the expense of
requiring operators to learn a new language and to 8. VOIP CLIENTS
mimic all the functionality which comes built into
Asterisk would require a very large learning curve. Clients and phones for use in VoIP networks are
To help Iptel the founders of SER provide many pre- now available as both hardware and software
built modules for plugging into SER including solutions; both being able to carry out the same
interface modules, accounting support and functionality of registering with a VoIP server to
voicemail. SERs primary intended use is as a SIP enable calls to be made to other clients. Wireless
proxy/router however also provides features to act handsets and mobile DECT technology phones are
as a registrar and redirect server. As a proxy/router also now available which are simply plugged into a
SER is designed to act as a standalone server and net cable at the base station rather than a phone
provides no functionality for direct communication cable. Although this technology is now becoming
with other servers. Redirection is built in, but this more widely available, users are still reluctant to buy
does not guarantee the user a connection and they into the hardware field as upgrade opportunities are
could just end up on a redirection loop or chain of limited without cost. Many solutions are at present
servers. Unlike Asterisk, SER has inbuilt support for reflecting the server technology they are designed
IPv6 as well as IPv4, and can listen for connections to connect to, the more advanced containing
on ports under both protocols concurrently. IPv6 answering machines and call holding ideal for use
capability provides greater support for mobile clients with a VOCAL server. Hardware phones are much
using the mobility headers of IPv6 and support for more commonly supporting the SIP protocol with
those users previously behind a NAT. SER provides H.323 support being relatively hard to find. This is
minimal extra interaction with users and support reflected in servers such as VOCAL where a
voicemail using a minimal set of base codecs as translations library for H.323 is provided as an add-
shown in Figure 5. on rather than a built-in.

6. Software solutions are much more variable in their as local deployment as a primary telephony service.
implementation and usage with upgrades being The project is looking at the many types of server
released at a constant rate to keep up with the technology and using both hardware and software
changing field and server technology. As with most clients to connect across a WAN with the additional
software many open source and free solutions can involvement of some willing volunteers from the
be found which install on any platform. Examples of Southampton Open Wireless Network (SOWN)6.
commercially produced clients include Windows Currently the system is operating on a single
Messenger1 for Windows and SJPhone2 for Asterisk server based in the main campus building
Windows and Linux. Neither of these handling all users and calling profiles. A SER server
implementations currently supports IPv6, and is also available on the same machine and is acting
Windows Messenger lacks DTMF (dial tone) as a proxy to Asterisk to provide IPv6 support for
generation facilities which prevent its use with testing with those clients which are able to use this
Voicemail and other touch-tone operated services. protocol. With SER acting as a proxy all extensions
Free and open source implementations include and user authentication is passed directly through to
KPhone3 and LinPhone4, both available for Linux. Asterisk by rewriting the incoming host port and
KPhone uses the KDE Qt library, while LinPhone translating the packets for use by Asterisk. This
has a GNOME GTK-based graphical interface. This solution operates without problems for calls placed
overview provides only a small selection from the between IPv4 users and IPv6 users providing both
ever expanding field where more companies are clients are using the same IP protocol. However if
getting involved on a daily basis. A recommended each client is using a different protocol the system
site to keep up with the latest is www.voip-info.org will fail on direct connection of the call due to the
which provides both listings of clients and server different connection types. To enable IPv4-IPv6
technology as well as useful guides to aid along the calls to be connected a further proxy has to be
way. provided to translate the packet headers to enable
each client to understand the data. This RTP-Proxy
As with the server technology the more popular
has to have support for both IPv4 and IPv6 traffic on
clients, software and hardware, mainly support the
the given network and the call between the clients
IPv4 protocol with limited support for IPv6. At the
should be routed through this proxy. Effectively an
time of writing IPv6 support in physical handsets
RTP-Proxy operates as a false client to which both
was still commercially unavailable. KPhone has
real clients send their information thinking that it is
been patched to support IPv6 using SER as the
their actual endpoint. The proxy then handles the
server, but this version has now become
traffic between the clients. This idea is good in
superseded and no longer works with most recent
prospect however research has shown not many of
release of SER and the Linux Kernel. LinPhone has
the client software packages currently support the
built in support for IPv6 and is being developed with
RTP-Proxy redirect information and still try to
this in mind, however this software implementation
connect the call directly [10]. This research is still on
is still at beta and contains many bugs [9].
going under the 6NET project being run worldwide
One of the most popular clients available currently including a section at the University.
operates on the Skype system which provides pure
VoIP using SIP (IPv4 only) through their own 10. CONCLUSION
software package. This has been designed with VoIP technology and distribution is on the increase
similar usage to MSN Messenger where a client is currently with many companies and institutes
able to have a phonebook of users who can be seen carrying out research in the area to further enhance
to be online or offline. This idea is now being this field, with the aim to provide the complete
expanded to include answering machine features solution. The two main protocols of SIP and H.323
and also to support various hardware phones now vary dramatically in their construction and usage of
becoming available with support for the Skype each has to be considered carefully. With the recent
system (www.skype.com). ruling by the FCC that SIP is not a specific
telephony protocol we now have a dramatic
9. VOIP IN OPERATION difference in the market between the two.
In this section a brief overview is offered as to the Governments are now backing the use of H.323 due
current state of research being performed at the to the legislation existing providing the ability to
University of Southampton into VoIP. This research monitor and control the networks usage. While
is being carried out in the Intelligence, Agents and 4 Windows Messenger -
Multimedia Group5 within the School of Electronics http://www.microsoft.com/windows/messenger/
and Computer Science and is focused on network 5 SJPhone – http://www.sjlabs.com
interoperability between IPv4 and IPv6 both over a 6 KPhone – http://www.iptel.org/products/kphone/
wired and wireless medium. The school regards 7 LinPhone – http://www.linphone.org
VoIP as an appropriate technology for roaming 8 IAM Research Group – http://www.iam.ecs.soton.ac.uk
academics (using mobility provided by IPv6) as well 9 SOWN – http://www.sown.org.uk

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8. 11. Appendix A
Codecs used in VoIP for communication between clients showing bit rate (quality) and bandwidth
consumption of each.
Standard bit rate sampling Raw Bandwidth
Name Description Remarks
by (kb/s) rate (kHz) Usage
(ADPCM
Intel, IMA ADPCM 32 8 var
) DVI
Also known as ulaw/alaw, mu-law
G.711 ITU-T Pulse code modulation (PCM) 64 8 87.2 Kbps
(US, Japan) and A-law (Europe)
Subband-codec that divides 16 kHz
G.722 ITU-T 7 kHz audio-coding within 64 kbit/s 64 16 * 120 Kbps + band into two subbands, each coded
using ADPCM
Coding at 24 and 32 kbit/s for hands-
G.722.1 ITU-T free operation in systems with low 24/32 16 * 60 Kbps + Variable Frame Size
frame loss
Dual rate speech coder for multimedia
Part of H.324 video conferencing.
G.723.1 ITU-T communications transmitting at 5.3 5.3/6.4 8 20.8/21.9 Kbps
DSP Group.
and 6.3 kbit/s
40, 32, 24, 16 kbit/s adaptive
16/24/32 31.5/47.2/55.2/6
G.726 ITU-T differential pulse code modulation 8 ADPCM; replaces G.721 and G.723.
/40 3.4 Kbps
(ADPCM)
5-, 4-, 3- and 2-bit/sample embedded
G.727 ITU-T adaptive differential pulse code var. ? var ADPCM. Related to G.726.
modulation (ADPCM)
Coding of speech at 16 kbit/s using
CELP. Annex J offers variable-bit
G.728 ITU-T low-delay code excited linear 16 8 31.5 Kbps
rate operation for DCME.
prediction
Coding of speech at 8 kbit/s using
G.729 ITU-T conjugate-structure algebraic-code- 8 8 31.2 Kbps Low delay (15 ms)
excited linear-prediction (CS-ACELP)
GSM Regular Pulse Excitation Long-Term
ETSI 13 8 30.3 Kbps Used for GSM cellular telephony.
06.10 Predictor (RPE-LTP)
10 coefficients. Also known as FIPS
LPC10e US Govt. Linear-predictive codec 2.4 8 7.8 Kbps
1015
iLBC (internet Low Bitrate Codec) Frames are encoded completely
iLBC IETF 13.3 8 27.7 Kbps
designed for narrow band speech. independently.
Speex is based on CELP and is
2.15-
Speex N/A designed to compress voice at bitrates 8/16/32 7.4 Kbps + open-source, multirate codec
44.2
ranging from 2 to 44 kbps.