Voip калькулятор и сжатие RTP

У Cisco есть страничка, где дается теория о расчете полосы для разных кодеков. Там же для зарегистрированных клиентов имеется ссылка на Voice Codec Bandwidth Calculator.

Полезные результаты по расчету полосы при использовании кодека g.729



Codec Bit Rate 8 kbps = (Codec Sample Size * 8) / (Codec Sample Interval)
Codec Sample Size 10 bytes size of each individual codec sample
Codec Sample Interval 10 msec the time it takes for a single sample

Codec: g729_All_Variants
Voice Payload Size: 20 bytes
Voice Protocol: VoIP
Compression: Not Applicable
Media Access: Ethernet
Tunnel/Security/Misc: None
Number of Calls: 1

Total Bandwidth (including Overhead) 32.76 kbps

Codec: g729_All_Variants
Voice Payload Size: 20 bytes
Voice Protocol: VoIP
Compression: off
Media Access: Frame-Relay
Tunnel/Security/Misc: None
Number of Calls: 1

Total Bandwidth (including Overhead) 28.14 kbps

Codec: g729_All_Variants
Voice Payload Size: 20 bytes
Voice Protocol: VoIP
Compression: on
Media Access: Frame-Relay
Tunnel/Security/Misc: None
Number of Calls: 1

Total Bandwidth (including Overhead) 12.18 kbps


Codec: g729_All_Variants
Voice Payload Size: 30 bytes
Voice Protocol: VoFR
Media Access: Not Applicable
Tunnel/Security/Misc: Not Applicable
Number of Calls: 1

Cisco IOS Total Bandwidth Needed for 1.0 Calls 10 kbps


Компрессия заголовков работает только на WAN интерфейсах с PPP,HDLC,FR-инкапсуляцией.
Пример:

interface Serial0/2/0:0.100 point-to-point
ip unnumbered Loopback0
frame-relay interface-dlci 100
frame-relay ip rtp header-compression

interface Serial0/1/0
ip unnumbered Loopback0
ip rtp header-compression
encapsulation ppp