o SIP and/or H.323 protocol used for call management
SIP currently supported only on Linux, H.323 supported on both Windows and Linux
SIP/UDP supported, H.323 uses both UDP and TCP
Can use directed mode, where VoIP phones call directly to themselves
Can also use Gateway mode where the VoIP phones register with a SIP or H.323 gateway
o RTP protocol used for streaming media transport, and supports the following CODECS (more codecs will be supported in the future):
G.711u: 64kbps data stream, 50 packets per second (SIP, H.323)
Speex: 16kbps data stream, 50 packets per second (SIP ONLY)
G.726-16: 16kbps data stream, 50 packets per second (SIP only)
G.726-24: 24kbps data stream, 50 packets per second (SIP only)
G.726-32: 32kbps data stream, 50 packets per second (SIP only)
G.726-40: 40kbps data stream, 50 packets per second (SIP only)
G.729a: 8kbps data stream, 50 packets per second (SIP only)
NONE: A messaging-only configuration is now supported (SIP only)
o Supports PESQ automated voice quality testing (not supported on Windows)
o RTCP protocol used for streaming media statistics (SIP only)
o Each LANforge VoIP/RTP endpoint can play from a wav file and record to a separate wav file (almost any sound file can be converted to the correct wav file format with tools bundled with LANforge - sample voice files included)
o Current benchmarks show support for 140 or more emulated VoIP phones per machine
o LANforge VoIP/RTP endpoints can call other LANforge endpoints or third-party SIP or H.323 phones like Cisco and Grandstream (third-party phones can also call LANforge endpoints and hear the WAV file being played)