*Network Requirements Commend

Specification Article | HTA-20210721-AW-01

Intercom |

Protocols:

IPv4 for configuration, IPv6 for VirtuoSIS
IoIP = UDP port 16384 (Communication between servers)
IoIP = UDP port 16400 (Communication between server and terminal)
TCP 17000 and 18000 (configuration)
SIP Port 5060 (by default)
SIPS (secure) Port 5061

Ethernet IEEE 802.3, 802.1p, 802.1q

IP Protocol (RFC 791)

ICMP (RFC 792)

UDP (RFC 768)

DTMF Signaling Method = RFC 2833

QoS / Diffserv RFC 2474
with Maximum one-way-delay of 100ms
Delay-Jitter not above 50ms
0% packet loss for perfect audio Quality.



 

Bandwidth

Depending on the number of configured Trunk connections to other Intercom Servers, the amount of signalling data increases. The following table shows the required minimum bandwidth for the network connection per Intercom Server location:

MTU

 G.711 is PCM—Based on the 8000 samples per second rate and 8 bits per sample, PCM generates 64,000 bits per second, or 64 Kbps. No compression is performed

G.722 is wideband speech encoding standard—G.722 divides the input signal into two subbands and encodes each subband using a modified version of ADPCM. G.722 supports a bit rate of 64 Kbps, 56 Kbps, or 48 Kbps.

DSP coverts analog voice signal to digital voice signal using a particular codec. Based on the codec used, the DSP generates so many bits per second. The bits that are generated for 10 milliseconds (ms) of analog voice signal form one digital voice sample. The size of the digital voice sample depends on the codec used. Table below shows how the digital voice sample size changes based on the codec used. The number of voice bytes for two digital voice samples using different codecs is shown in the last column.

https://www.ccexpert.us/ont/impact-of-voice-samples-and-packet-size-on-bandwidth.html

 

SIP

SIP transport over UDP port 5060 (can be configured) following RFC 3261 and RFC 3265.
DTMF support via SIP Info, Inband or RFC 2833.
IAX2 support following RFC 5456.
RTP port ranges for audio or video transport can be configured.
Preferred audio codecs are G.711 a/u-law.
Other Codecs like G.722, Speex, GSM, iLBC, aso. are supported or can be transcoded.

Intercom Server Configuration manual: PM-Pro800-Basic-EN.fm (commend.com)