Asterisk Documentation 1.4 chaniax.txt

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Inter-Asterisk eXchange Protocol


This document is intended as an introduction to the Inter-Asterisk 
eXchange (or simply IAX) protocol.  It provides both a theoretical 
background and practical information on its use.

The first question most people are thinking at this point is "Why do you 
need another VoIP protocol?  Why didn't you just use SIP or H.323?"

Well, the answer is a fairly complicated one, but in a nutshell it's like
this...  Asterisk is intended as a very flexible and powerful
communications tool.  As such, the primary feature we need from a VoIP
protocol is the ability to meet our own goals with Asterisk, and one with
enough flexibility that we could use it as a kind of laboratory for
inventing and implementing new concepts in the field.  Neither H.323 or
SIP fit the roles we needed, so we developed our own protocol, which,
while not standards based, provides a number of advantages over both SIP
and H.323, some of which are:

	* Interoperability with NAT/PAT/Masquerade firewalls
	     IAX seamlessly interoperates through all sorts of NAT and PAT
             and other firewalls, including the ability to place and 
             receive calls, and transfer calls to other stations.

	* High performance, low overhead protocol
	     When running on low-bandwidth connections, or when running 
	     large numbers of calls, optimized bandwidth utilization is 
	     imperative.  IAX uses only 4 bytes of overhead

	* Internationalization support
	     IAX transmits language information, so that remote PBX 
	     content can be delivered in the native language of the
	     calling party.

	* Remote dialplan polling
	     IAX allows a PBX or IP phone to poll the availability of a 
	     number from a remote server.  This allows PBX dialplans to 
	     be centralized.

	* Flexible authentication
	     IAX supports cleartext, md5, and RSA authentication, 
	     providing flexible security models for outgoing calls and 
	     registration services.
	* Multimedia protocol
	     IAX supports the transmission of voice, video, images, text, 
	     HTML, DTMF, and URL's.  Voice menus can be presented in both
	     audibly and visually.

	* Call statistic gathering
	     IAX gathers statistics about network performance (including 
	     latency and jitter, as well as providing end-to-end latency

	* Call parameter communication
	     Caller*ID, requested extension, requested context, etc are
	     all communicated through the call.

	* Single socket design
	     IAX's single socket design allows up to 32768 calls to be 
While we value the importance of standards based (i.e. SIP) call handling, 
hopefully this will provide a reasonable explanation of why we developed 
IAX rather than starting with SIP.

Lines beginning with '>' represent lines which might appear in an actual 
configuration file.  The '>' is used to help separate them from the 
descriptive text and should not actually be included in the file itself.

Lines within []'s by themselves represent section labels within the 
configuration file.  like this:

> [mysection]

Options are set using the "=" sign, for example

> myoption = value

Sometimes an option will have a number of discrete values which it can 
take.  In that case, in the documentation, the options will be listed 
within square brackets (the "[" and "]" ones) separated by the pipe symbol 
("|").  For example:

> myoption = [value1|value2|value3]

means the option "myoption" can be assigned a value of "value1", "value2", 
or "value3".

Objects, or pseudo-objects are instantiated using the "=>" construct.  For 

> myobject => parameter

creates an object called "myobject" with some parameter whose definition
would be specific to that object.  Note that the config file parser
considers "=>" and "=" to be equivalent and their use is purely to make
configuration files more readable and easier to "humanly parse".

The comment character in Asterisk configuration files is the semicolon 
";".  The reason it is not "#" is because the "#" symbol can be used as 
parts of extensions and it didn't seem like a good idea to have to escape 


Like everything else in Asterisk, IAX's configuration lies in 
/etc/asterisk -- specifically /etc/asterisk/iax.conf

The IAX configuration file is a collection of sections, each of which
(with the exception of the "general" section) represents an entity within 
the IAX scope.


The first section is typically the "general" section.  In this area, 
a number of parameters which affect the entire system are configured.  
Specifically, the default codecs, port and address, jitter behavior, TOS 
bits, and registrations.

The first line of the "general" section is always:

> [general]

Following the first line are a number of other possibilities:

> bindport = <portnum>

This sets the port that IAX will bind to.  The default IAX version 1 
port number is 5036.  For IAX version 2, that is now the default in
Asterisk, the default port is 4569.
It is recommended that this value not be altered in general.

> bindaddr = <ipaddr>

This allows you to bind IAX to a specific local IP address instead of
binding to all addresses.  This could be used to enhance security if, for
example, you only wanted IAX to be available to users on your LAN.

> bandwidth = [low|medium|high]

The bandwidth selection initializes the codec selection to appropriate
values for given bandwidths.  The "high" selection enables all codecs and
is recommended only for 10Mbps or higher connections.  The "medium"
bandwidth eliminates signed linear, Mu-law and A-law codecs, leaving only
the codecs which are 32kbps and smaller (with MP3 as a special case).  It
can be used with broadband connections if desired. "low" eliminates ADPCM
and MP3 formats, leaving only the G.723.1, GSM, and LPC10.

> allow = [gsm|lpc10|g723.1|adpcm|ulaw|alaw|mp3|slinear|all]
> disallow = [gsm|lpc10|g723.1|adpcm|ulaw|alaw|mp3|slinear|all]

The "allow" and "disallow" allow you to fine tune the codec selection 
beyond the initial bandwidth selection on a codec-by-codec basis.  

The recommended configuration is to select "low" bandwidth and then 
disallow the LPC10 codec just because it doesn't sound very good. 

> jitterbuffer = [yes|no]
> dropcount = <dropamount>
> maxjitterbuffer = <max>
> maxexcessbuffer = <max>

These parameters control the operation of the jitter buffer.  The 
jitterbuffer should always be enabled unless you expect all your 
connections to be over a LAN.  
* drop count is the maximum number of voice packets to allow to drop 
  (out of 100).  Useful values are 3-10.  
* maxjitterbuffer is the maximum amount of jitter buffer to permit to be 
* maxexcessbuffer is the maximum amount of excess jitter buffer 
  that is permitted before the jitter buffer is slowly shrunk to eliminate 
* minexcessbuffer is the minimum amount of excess jitter buffer

> accountcode = <code>
> amaflags = [default|omit|billing|documentation]

These parameters affect call detail record generation.  The first sets the 
account code for records received with IAX.  The account code can be 
overridden on a per-user basis for incoming calls (see below).  The 
amaflags controls how the record is labeled ("omit" causes no record to be 
written.  "billing" and "documentation" label the records as billing or 
documentation records respectively, and "default" selects the system 

> tos = [lowdelay|throughput|reliability|mincost|none]

IAX can optionally set the TOS (Type of Service) bits to specified values 
to help improve performance in routing.  The recommended value is 
"lowdelay", which many routers (including any Linux routers with 2.4 
kernels that have not been altered with ip tables) will give priority to 
these packets, improving voice quality.

> register => <name>[:<secret>]@<host>[:port]

Any number of registry entries may be instantiated in the general 
section.  Registration allows Asterisk to notify a remote Asterisk server 
(with a fixed address) what our current address is.  In order for 
registration to work, the remote Asterisk server will need to have a 
dynamic peer entry with the same name (and secret if provided).  

The name is a required field, and is the remote peer name that we wish to 
identify ourselves as.  A secret may be provided as well.  The secret is 
generally a shared password between the local server and the remote 
server.  However, if the secret is in square brackets ([]'s) then it is 
interpreted as the name of a RSA key to use.  In that case, the local Asterisk 
server must have the *private* key (/var/lib/asterisk/keys/<name>.key) and 
the remote server will have to have the corresponding public key.

The "host" is a required field and is the hostname or IP address of the 
remote Asterisk server.  The port specification is optional and is by 
default 4569 for iax2 if not specified.

> notransfer = yes | no

If an IAX phone calls another IAX phone by using a Asterisk server, 
Asterisk will transfer the call to go peer to peer. If you do not
want this, turn on notransfer with a "yes". This is also settable
for peers and users.


The following sections, after "general" define either users, peers or
friends.  A "user" is someone who connects to us.  A "peer" is someone
that we connect to.  A "friend" is simply shorthand for creating a "user" 
and "peer" with identical parameters (i.e. someone who can contact us and 
who we contact). 

> [identifier]

The section begins with the identifier in square brackets.  The identifier 
should be an alphanumeric string.

> type = [user|peer|friend]

This line tells Asterisk how to interpret this entity.  Users are things 
that connect to us, while peers are phones we connect to, and a friend is 
shorthand for creating a user and a peer with identical information

User fields:

> context = <context>

One or more context lines may be specified in a user, thus giving the user 
access to place calls in the given contexts.  Contexts are used by 
Asterisk to divide dialing plans into logical units each with the ability 
to have numbers interpreted differently, have their own security model, 
auxiliary switch handling, and include other contexts.  Most users are 
given access to the default context.  Trusted users could be given access 
to the local context for example.

> permit = <ipaddr>/<netmask>
> deny = <ipaddr>/<netmask>

Permit and deny rules may be applied to users, allowing them to connect 
from certain IP addresses and not others.  The permit and deny rules are 
interpreted in sequence and all are evaluated on a given IP address, with 
the final result being the decision.  For example:

> permit =
> deny =

would deny anyone in with a netmask of 24 bits (class C), 

> deny =
> permit =

would not deny anyone since the final rule would permit anyone, thus 
overriding the denial.  

If no permit/deny rules are listed, it is assumed that someone may connect 
from anywhere.

> callerid = <callerid>

You may override the Caller*ID information passed by a user to you (if 
they choose to send it) in order that it always be accurate from the 
perspective of your server.

> auth = [md5|plaintext|rsa]

You may select which authentication methods are permitted to be used by 
the user to authenticate to us.  Multiple methods may be specified, 
separated by commas. If md5 or plaintext authentication is selected, a 
secret must be provided. If RSA authentication is specified, then one or 
more key names must be specified with "inkeys"

If no secret is specified and no authentication method is specified, then 
no authentication will be required.

> secret = <secret>

The "secret" line specifies the shared secret for md5 and plaintext 
authentication methods.  It is never suggested to use plaintext except in 
some cases for debugging.

> inkeys = key1[:key2...]

The "inkeys" line specifies which keys we can use to authenticate the 
remote peer.  If the peer's challenge passes with any of the given keys, 
then we accept its authentication.  The key files live in 
/var/lib/asterisk/keys/<name>.pub and are *public keys*.  Public keys are 
not typically DES3 encrypted and thus do not usually need initialization.

Peer configuration

> allow = [gsm|lpc10|g723.1|adpcm|ulaw|alaw|mp3|slinear|all]
> disallow = [gsm|lpc10|g723.1|adpcm|ulaw|alaw|mp3|slinear|all]

The "allow" and "disallow" may be used to enable or disable specific codec 
support on a per-peer basis.  

> host = [<ipaddr>|dynamic]

The host line specifies the hostname or IP address of the remote host, or 
may be the word "dynamic" signifying that the host will register with us 
(see register => in the general section above).

> defaultip = <ipaddr>

If the host uses dynamic registration, Asterisk may still be given a 
default IP address to use when dynamic registration has not been performed 
or has timed out.

> peercontext = <context>

Specifies the context name to be passed to the peer for it to use when routing
the call through its dial plan. This entry will be used only if a context
is not included in the IAX2 channel name passed to the Dial command.

> qualify = [yes | no | <value>]

Qualify turns on checking of availability of the remote peer. If the 
peer becomes unavailable, no calls are placed to the peer until
it is reachable again. This is also helpful in certain NAT situations.

> jitterbuffer = [yes | no]

Turns on or off the jitterbuffer for this peer

> mailbox = <mailbox>[@mailboxcontext]

Specifies a mailbox to check for voicemail notification.

> permit = <ipaddr>/<netmask>
> deny = <ipaddr>/<netmask>

Permit and deny rules may be applied to users, allowing them to connect 
from certain IP addresses and not others.  The permit and deny rules are 
interpreted in sequence and all are evaluated on a given IP address, with 
the final result being the decision.  See the user section above 
for examples.

For more examples of a configuration, please see the iax.conf.sample in
your the /configs directory of you source code distribution

Created by: josiahbryan, Last modification: Sun 25 of Jul, 2010 (08:37 UTC)
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