Setup guide

• Quicknet Technologies cards:
• Internet PhoneJACK (ISA or PCI) for connecting an analog telephone (FXS port)
• Internet LineJACK (ISA) for connecting an analog telephone line (FXO port) or a
telephone (FXS port)
• ISDN client cards (PCI) for connecting an ISDN line. See Device Driver List for the list of
supported PCI ISDN cards
• Voicetronix OpenLine4 card for connecting four (4) analog telephone lines (FXO ports)
• Zaptel Wildcard X100P IP telephony card (from Linux Support Services) for connecting one
analog telephone line (FXO port)
Supported standards:
• Wandy RouterOS supports IP Telephony in compliance with the International
Telecommunications Union - Telecommunications (ITU-T) specification H.323v4. H.323 is a
specification for transmitting multimedia (voice, video, and data) across an IP network.
H.323v4 includes: H.245, H.225, Q.931, H.450.1, RTP(real-time protocol)
• The followong audio codecs are supported: G.711 (the 64 kbps Pulse code modulation (PCM)
voice coding), G.723.1 (the 6.3 kbps compression technique that can be used for compressing
audio signal at very low bit rate), GSM-06.10 (the 13.2 kbps coding), LPC-10 (the 2.5 kbps
coding), G.729 and G.729a (the 8 kbps CS-ACELP software coding), G.728 (16 kbps coding
technique, supported only on Quicknet LineJACK cards)
In PSTN lines there is a known delay of the signal caused by switching and signal compressing
devices of the telephone network (so, it depends on the distance between the peers), which is
generally rather low. The delay is also present in IP networks. The main difference between a PSTN
and an IP network is that in IP networks that delay is more random. The actual packet delay may
vary in order of magnutude in congested networks (if a network becomes congested, some packets
may even be lost). Also packet reordering may take place. To prevent signal loss, caused by rendom
jitter of IP networks and packet reordering, to corrupt audio signal, a jitter buffer is present in IP
telephony devices. The jitter buffer is delaying the actual playback of a received packet forming
The larger the jitter buffer, the larger the total delay, but fewer packets get lost due to timeout.
The total delay from the moment of recording the voice signal till its playback is the sum of
following three delay times:
• delay time at the recording point (approx. 38ms)
• delay time of the IP network (1..5ms and up)
• delay time at the playback point (the jitter delay)
Notes
Each installed Quicknet card requires IO memory range in the following sequence: the first card
occupies addresses 0x300-0x31f, the second card 0x320-0x33f, the third 0x340-0x35f, and so on.
Make sure there is no conflict in these ranges with other devices, e.g., network interface cards, etc.
Use the telephony logging feature to debug your setup.
Additional Documents
General Voice port settings
ip telephony voice-port