SIP Participants
This chapter only deals 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
In principle, 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 only possible if Antcas is configured as server. |
| Mobotix T2X and S16, S26 | |
| Rene Koch | For this system, the behavior on incoming call must always be busy, otherwise simultaneous use of the Rene Koch app is not possible. |
| Siedle with SIP Gateway | |
| SIP Provider | To integrate a 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 an Telephone in the
Library for each device and service. Telephones are only registered when a visualization port is assigned to the visualization. An
Telephone can be configured as server, telephone or both. Configuration is generally simpler if the telephone registers with the counterpart. However, not all counterparts allow this.
Note: Ensure that there are enough
SIP Participant Licenses. Otherwise, the telephone will not be configured and is not reachable. Each telephone plus any function that makes a call requires a license. This can be an outgoing call or a self-call.
The created Telephone uses a dial plan to define the function. First, telephone registration should be implemented. The dial plan can then be modified accordingly. This is further explained in the corresponding subchapter Dial Plan.
A SIP address defines the phone number of a participant. For example, this is configured as 100@10.0.0.200, 100@server.tld or 100. The local counterpart usually uses the format, for example, 100@10.0.0.200. The target number is defined in the function Dial. This also corresponds to the SIP format. An optional port can be specified here, for example, 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 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. Then, a participant's registration is configured. Under Information, the status of the registration is displayed. Note that this status may change with delay depending on 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, for example, displayed as the 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 target 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. Then, the participant's settings are configured. Under Information, the status is displayed. Note that this status may change with delay depending on 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. Target number: id@server | projectid_telephoneid |
| Name | The call identifier. This field is optional and, for example, displayed as the 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 participant registration and communication. The domain can be configured in 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