SIP Participants
In this chapter, we only deal with the configuration of communication with an external device or service. The extended configuration of visualization is further explained in the chapter Visualization (WebRTC).
Supported Devices And Services
Basically, all systems can be integrated that fully support the SIP protocol. Important is the support of RFC 3261 Section 19.1.4 (URI Comparison), which allows case-sensitive alphanumeric addresses if the device must be integrated as a telephone.
| Manufacturer | Remarks |
|---|---|
| 2N | |
| Commend | |
| Doorbird | Integration is only possible if Antcas is configured as a server. |
| Mobotix T2X and S16, S26 | |
| Rene Koch | With this system, the behavior on incoming calls must always be busy, otherwise simultaneous use of the Rene Koch app is not possible. |
| Siedle with SIP Gateway | |
| SIP Provider | To integrate an SIP provider, a domain may need to be configured. This can only be used across all participants. |
Configuration
Before configuration can begin, a participant must be created as a Telephone in the
Library for each device and service. Telephones are only registered when a visualization port has been assigned to the visualization. An
Telephone can be configured as a server, telephone or both. Basically, configuration is simpler if the telephone registers with the counterpart. However, not all counterparts allow this.
Note: Make sure that there are enough
SIP Participant Licenses. Otherwise, the telephone will not be configured and will not be reachable. Each telephone is license-bound, plus every function that makes a call. This can be an outgoing call or a self-call.
The created Telephone uses a dial plan to define the function. First, the registration of the telephone should be implemented. After that, the dial plan can be modified accordingly. This is further explained in the corresponding subchapter Dial Plan.
A SIP address defines the phone number of a participant. This is configured e.g. as 100@10.0.0.200, 100@server.tld or 100. The local counterpart usually uses the format e.g. 100@10.0.0.200. The destination number is defined in the function Dial. This also corresponds to the SIP format. An optional port can be specified here, e.g. 100@10.0.0.200:5060.
The configuration of a participant can take on various types. The following table shows the differences.
| Type | Description |
|---|---|
| Telephone | A telephone registers as a client with a VOIP server or SIP provider. |
| Server | Allows an endpoint to register on the Antcas server. |
| Server and Telephone | This method defines a participant as both a server and telephone. This method is not recommended. |
| Disabled | Disables the telephone. This is useful to temporarily shut down a system or manage templates. |
Configuration As Telephone
On the right side of the Telephone, select the type Telephone. After that, the registration of a participant is configured. Under Information, the status of the registration is displayed. Note that this status may change with delay depending on the timeout. If authentication fails, this will not be displayed. The following table shows further details of the configuration:
| Name | Value | Example |
|---|---|---|
| Type | Telephone | |
| Participant ID | Can be used to trigger the server by call. Phone number: id@server | projectid_telephoneid |
| Name | The call identifier. This field is optional and is displayed e.g. as a name on the telephone. | Front Door |
| SIP Address | Defines the phone number of the participant. | 100@participantip |
| Username | Username for authentication. | user |
| Optional Username | This username is used for authentication by SIP providers. It is not used for intercom systems or local telephones. | |
| Password | Password for registration. | wtqhZ25j0HgJqQq4 |
| Server | Target server for registration and dialing with authentication. This must be the same as that of the selected destination number. | participantip |
| Port | UDP target port | 5'060 or 5'061 |
Configuration As Server
On the right side of the Telephone, select the type Server. After that, the settings of the participant are configured. Under Information, the status is displayed. Note that this status may change with delay depending on the timeout. If authentication fails, this will not be displayed. The following table shows further details of the configuration:
| Name | Value | Example |
|---|---|---|
| Type | Server | |
| Participant ID | The username for login. Can also be used to trigger the server by call. Destination number: id@server | projectid_telephoneid |
| Name | The call identifier. This field is optional and is displayed e.g. as a name on the telephone. | Front Door |
| SIP Address | Defines the phone number of the server. | 100@serverip |
| Username | Username for authentication when dialing. This should ideally be identical to the participant ID. | Participant-ID |
| Password | Password for dialing and logging in to the server. | wtqhZ25j0HgJqQq4 |
Domain
The use of the domain enables external services to be used. Antcas Control only supports a global domain across all projects. This domain is used for the registration and communication of participants. The domain can be configured in the Infrastructure under
Network
Telephony.
Codecs
The following codecs are available for communication:
- Opus (only from Core 2 and higher)
- G.722
- PCMU ITU G.711 ulaw
- PCMA ITU G.711 alaw