Link layer: MAC address: XX:XX:XX:XX:XX:XX. 48 bits (half of them brand number, the other half random). Theoretically it is fixed… but users can manipulate it since it is managed by the operative system.
Network layer: IPv4 address: X.X.X.X.
Transport layer: Two different protocols:
TCP: receipt acknowledgement. Slow. Used for almost everything.
UDP: no receipt acknowledgement. Fast. Used for all streaming technologies (video on demand, online radio, music streaming… and VoIP)
The transport layer uses port numbers. Examples: http: 80, ssh: 22, smb:445…
0 - 1023: Reserved for standard applications.
1024 - 49151: Reserved for private applications.
49152 - 65535: Not reserved. “Dynamic ports”.
Some examples: SIP uses port UDP 5060 for signaling (others for voice packets), IAX uses UDP 4569 for voice and signaling.
VoiP uses UDP, but it has means to compensate jitter1. RTP “Real Time Protocol” sets a timestamp indicating the max waiting time before ditching the packet. RTCP (RTP Control Protocol) adjusts speed.
Network parameters
A network interface has to have these parameters:
IP address. Remember: Private IP addresses (not addressable from the internet except when using NAT):
10.X.X.X
172.16.0.0 - 172.31.255.255
192.168.X.X
Gateway: IP address of the device where packets bound to other networks are sent.
Subnet mask. It separates the network part from the host part. Formats:
Decimal (used in Windows and macOS): 255.255.255.0
Binary (never used, only for calculations): 11111111.11111111.11111111.00000000
Prefix lenght (used in Linux, much more clear): /24
DNS servers: They translate web addresses to IP addresses.
NAT is a technique used by CPE routers to save IP addresses. It encapsulates the packets using the router’s public IP address. SIP has problems when dialing extensions in different networks.
SIP clients
Apart from all the aforementioned (IP, netmask…), SIP clients (softphones or physical phones) have to include these parameters:
Extension number / username.
Password / secret.
PABX IP address / domain / SIP Proxy. IP address of the SIP server (Elastix/Freepbx/Physical PBX… Asterisk machine)
Contrary to popular belief, SIP clients don’t need static IP addresses, their basic network parameters (IP + netmask + GW + DNS) can be assigned via DHCP.
IP disasters
There are times when you have to configure a device with a wrong network configuration. Sometimes without possible offline serial RS-232 cable. What are you supposed to do…?
Interesting network features
QoS: “Quality of Service”. Network devices (routers and switches) detect VoIP traffic and prioritise it (reserve bandwidth) over other kinds of traffic.
VLAN: Separate VoIP traffic in different network. You will know next year.
VoIP signaling protocols
Signaling: network traffic standard used to login users/extensions into the PABX and to manage telephony aspects: tone, protocol used to dial…
Main signaling protocols:
SIP. Most used.
IAX (Inter-Asterisk Exchange). Better and more simple than SIP. But it arrived too late.
H.323
Other privative systems: Skype, Hangouts…
VoIP Audio codecs (no .mp3 here)
Remember: codec = encoder-decoder. Codecs determine the audio system and compression algorithm used in the voice stream.
G.711 (also used by RDSI)
DPCM
ADPCM
LPC
CELP
G.729 (used under license) Very good quality on low bitrate (8 kbps)
G.723.1
GSM
VSELP
AMR
iLBC
SILK
Can you tell the difference?
Asterisk: It is a free software application.
Elastix: It is a Linux distribution with a nice GUI (Graphic User Interface) that includes Asterisk. It is a commercial distribution (not free). There are others with free software such as FreePBX.
Physical SIP PABX also exist. We will meet them in the next term.
