====== Voice support ======
The NetModule Router Series NB2000 and NB3000 support voice and audio communication with two extension:

  - **Voice-License ("V")** GSM-Voice to VoIP/SIP to make or receive PSTN voice call over the mobile network.
  - **Audio ("A")** stereo line-in/out to VoIP/SIP to connect audio signals directly to the device. Typical microphone and loudspeaker via external amplifier.
  - **Push-to-Talk ("Ap")** mono line in/out to connect audio signals directly to the device, 1/1 digital in/out e.g. for call & ring signaling


===== Basic concept =====
An endpoint is a SIP/VoIP device like
  * Internal GSM-Voice
  * Internal Audio-Interface
  * External SIP-Phone, PBX, door station etc.

To establish a call two steps are needed:
  - Signalling  = establish a call
  - Call-Routing = making the connection
Simple behaviour can be configured by the Web-GUI. 
For more complex applications the SDK can be used.

{{:app-notes:voice1.jpg?600|}}

===== Voice gateway configuration via Web GUI=====

Using the router with a SIP phone client.
==== Administration ====
{{:app-notes:voice1.png?600|}}
==== Endpoints ====
{{:app-notes:voice2.png?600|}

^ Parameter ^ Voice Gateway Endpoint Types ^
| Voice-Over-Mobile | Endpoint for GSM/UMTS/LTE calls (can be used for calls to mobile or landline phones)|
| SIP (registrar)| SIP endpoint which can be a client registered to our registrar |
| SIP (direct)| Endpoint for calls directly routed to a SIP agent without registration |
| SIP (user-agent) | Endpoint acting as SIP user agent towards a remote registrar |
==== Endpoint types ====
{{:app-notes:voice3.jpg?600|}}
==== Routing ====
{{:app-notes:voice4.png?600|}}
==== Routing modes ====
{{:app-notes:voice5.png?600|}}

===== SDK scripts =====
Endpoints are represented as structs
<code>
struct(3): {
	.id = int: 54321
	.desc = string[5]: "vom://Vom1"
	.state = string[4]: "busy"
	.volume = = int: 7
};
</code>

Descriptors
  * Sip1 First SIP subscriber
  * Vom1 First Voice-Over-Mobile
  * Aud1 First Audio device
  * sip:⁄⁄user@192 .168.1.254:5060 SIP address

States
  * busy endpoint is already holding a call
  * available endpoint is ready to take a call

Calls are represented as structures which may look like:
<code>
struct(5): {
	.id = int: 12345
	.state = string[7]: "dialing"
	.calling = string[24]: "sip://user@192 .168.1.254:5060"
	.called = string[22]: "vom://+123456789 @Vom1"
};
</code>

The following states are possible:
  * routing call is in routing state
  * dialing call is in dialing state
  * alerting call is in alerting state
  * active call is active
  * hungup call had hung up

Voice functions:
  * array nb_voice_endpoint_list (void) / struct nb_voice_endpoint_get (endpoint) 

  * Unordered List Itemarray nb_voice_call_list (void) / struct nb_voice_call_get (call)

  * int nb_voice_call_dial (call) 
  * int nb_voice_call_accept (call)
  * int nb_voice_call_route (call, endpoint)
  * int nb_voice_call_hangup (call)

  * int nb_voice_call_volume (endpoint, int level)

==== How to make a call ====
=== Start Call ===
  * Unordered List Itemnb_voice_call_dial(call) or wait for call
=== Event magement ===
  * Unordered List ItemOutgoing – do nothing
  * Incomming - nb_voice_call_route(call, target endpoint)
  * Dispatched: alerting ->  nb_voice_call_accept(call)
=== End Call ===
  * nb_voice_call_hangup(call)

[[sdk:scripts:voice-dispatcher-audio|Sample SDK Script voice call dispatching]]

----

[[configuration:configuration|Example: Voice Gateway Configuration File]]