June 2015 - Ed.2.1
Audio over IP decoders E411, E413, E415 & E417 technical manual
Page 22
Multicast: The device will aggregate itself to the group on the configured multicast IP address and will
receive RTP packets on the selected port.
The selected port must be even and not used for another channel in the device.
‘SIP PBX’ Source type configuration
In this case, the audio communication is established with a SIP device (i.e. a telephone) through a SIP PBX.
The codec used is G711-A (also known as PCMA). This is the most common codec used in VoIP systems
and almost 100% of the devices support it.
The device registers itself as an extension on the PBX and answers automatically the incoming calls,
activating the channel. The channel is deactivated when the call is ended by the other end (telephone).
The configuration form for this source is:
Figure 17: SIP PBX source type setup form
The parameters to configure are:
PBX IP: IP address of the PBX.
PBX port: SIP port of the PBX (the standard is 5060).
Refresh time: Duration of the registration. The device must renew the registration in the PBX with this
periodicity (time in seconds).
User: The extension assigned to the device in the PBX.
Password: Security password configured in the PBX for the assigned extension.
Local RTP port: Port for the audio RTP. Must be an even number between 0 and 65534, and not used for
other channels in the device.
