AVT MAGIC AC1 Go Configuration Manual page 9

▪ Establish a connection
by dialling a phone number via a SIP server
▪
by entering the IP address of the receiver via Direct
▪
SIP
Support for up to 5 SIP provider profiles which are
▪
active simultaneously
Uses SDP to negotiate the coding algorithm
▪
Audio transmission via RTP over UDP
▪
Numbered packets allow for packet loss detection
▪
▪ The packet size is variable if it is not determined
by the codec used.
▪ Max. buffer size: 2 seconds
▪ Mitigation of lost packets by repeating packets
on the receiving side
AoIP/SIP
▪ Support for the STUN protocol (Session Traversal
Utilities for NAT)
Simplified: Conversion of the internal IP address
▪
into an external IP address
▪ Supports Quality of Service (DiffServ) to prioritize
audio and SIP independently
▪ Interworking with SIP telephones, in SD quality or
HD-Voice
▪ Secure Streaming Function
Sending two streams with identical content
▪
▪ A delay between the streams can be configured
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