Asterisk readme iax

A copy of the README.IAX in the Asterisk distribution as of 09/22/2003:


Inter-Asterisk eXchange Protocol

INTRODUCTION



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.

WHY IAX


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 imperitive. 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 measurement.
  • 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 multiplexed.

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.

CONFIG FILE CONVENTIONS



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 example:
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 it.

IAX CONFIGURATION IN ASTERISK



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:

  • port = <portnum>: This sets the port that IAX will bind to. The default IAX port number is 5036. 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>: Sets a hard limit for the jitterbuffer (value in ms).
  • 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. The drop count is the maximum number of voice packets to allow to drop (out of 100). Useful values are 3-10. The maxjitterbuffer is the maximum amount of jitter buffer to permit to be used. The "maxexcessbuffer" is the maximum amount of excess jitter buffer that is permitted before the jitter buffer is slowly shrunk to eliminate latency.
  • 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 default. (See Asterisk billing.
  • 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 registery 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 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 5036 if not specified.


IAX Client configuration


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 people 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, auxilliary 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 = 0.0.0.0/0.0.0.0
deny = 192.168.0.0/255.255.255.0

would deny anyone in 192.168.0.0 with a netmask of 24 bits (class C),
whereas:

deny = 192.168.0.0/255.255.255.0
permit = 0.0.0.0/0.0.0.0

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 specifed 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.



See also

A copy of the README.IAX in the Asterisk distribution as of 09/22/2003:


Inter-Asterisk eXchange Protocol

INTRODUCTION



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.

WHY IAX


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 imperitive. 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 measurement.
  • 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 multiplexed.

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.

CONFIG FILE CONVENTIONS



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 example:
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 it.

IAX CONFIGURATION IN ASTERISK



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:

  • port = <portnum>: This sets the port that IAX will bind to. The default IAX port number is 5036. 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>: Sets a hard limit for the jitterbuffer (value in ms).
  • 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. The drop count is the maximum number of voice packets to allow to drop (out of 100). Useful values are 3-10. The maxjitterbuffer is the maximum amount of jitter buffer to permit to be used. The "maxexcessbuffer" is the maximum amount of excess jitter buffer that is permitted before the jitter buffer is slowly shrunk to eliminate latency.
  • 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 default. (See Asterisk billing.
  • 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 registery 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 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 5036 if not specified.


IAX Client configuration


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 people 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, auxilliary 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 = 0.0.0.0/0.0.0.0
deny = 192.168.0.0/255.255.255.0

would deny anyone in 192.168.0.0 with a netmask of 24 bits (class C),
whereas:

deny = 192.168.0.0/255.255.255.0
permit = 0.0.0.0/0.0.0.0

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 specifed 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.



See also

Created by: oej, Last modification: Wed 18 of Jul, 2007 (11:29 UTC) by swatchy
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