June 2015 - Ed.2.1
Audio over IP decoders E411, E413, E415 & E417 technical manual
Page 23
Buffer size: number of packets in the input buffer. The greater the value, the more network jitter the
decoder is able to absorb, but the greater the delay in relation to the emitter. Default value is 8.
‘P2P SIP’ Source type configuration
This source type is similar to the PBX SIP one but, in this case, the communication is established using the
SIP protocol directly between two endpoints without using a PBX. This method to communicate SIP
devices is known as “peer to peer” or “P2P”.
The absence of a PBX forces the calling device (usually a telephone) to know the IP address of the unit
called (in this case the E41x decoder) and make the call using this IP address number instead a simple
numeric extension.
IMPORTANT NOTE:
This is not possible with all the phones in the market. So, special care has to be taken
when choosing the phone used to make the calls, and also with its configuration.
The configuration form for this source type is:
Figure 18: SIP P2P source type setup form
The parameters to be set are:
Local extension: The extension assigned to the device. Even it cannot be used to call the device in P2P
mode, is mandatory for the SIP protocol to have an extension assigned for each device. You may use any
combination of numbers and letters up to 25 characters. Spaces are not allowed.
Local RTP port: Port for receiving the RTP audio packets. Must be an even number between 0 and 65534
and not be used for other channels in the device.
Buffer size: number of packets in the input buffer. The greater the value, the more network jitter the
decoder is able to absorb, but the greater the delay in relation to the emitter. Default value is 8.
