VoIP & SIP
VoIPAlso known as Voice over Internet Protocol, IP telephony
Definition
VoIP (Voice over Internet Protocol) is the delivery of voice calls over IP networks instead of dedicated circuit-switched telephone lines. Speech is digitised by a codec, split into small packets, sent across the internet or a private network, and reassembled at the far end, with a signalling protocol such as SIP setting up and ending each call.
How VoIP works
A VoIP call has two separate flows. Signalling sets up, modifies and tears down the call; today this is almost always SIP (RFC 3261), with H.323 still found in older systems. Media carries the audio itself as RTP packets (RFC 3550), usually one packet every 20 ms.
Before sending, a codec compresses the audio. G.711 uses 64 kbit/s and sounds like a normal landline; G.729 squeezes speech into 8 kbit/s; wideband codecs such as G.722 and Opus capture more of the voice spectrum for “HD voice”.
What affects VoIP call quality
Because packets share the network with other traffic, quality depends on three measurable things:
- Latency: ITU-T G.114 recommends keeping one-way mouth-to-ear delay under about 150 ms for natural conversation.
- [Jitter](/glossary/jitter): variation in packet arrival time, absorbed by a jitter buffer at the receiver.
- Packet loss: lost packets cause gaps or robotic audio; above roughly 1–2 % most listeners notice.
NAT traversal problems are the other classic cause of trouble, typically showing up as one-way audio.
VoIP and mobile gateways
VoIP systems reach the public phone network through SIP trunks from carriers, or through gateways that bridge to other networks. A GSM gateway bridges VoIP to mobile lines. GSMCalls manages phones as mobile SIP gateways and monitors latency, loss and jitter per trunk and device from the platform.
Frequently asked questions
Is VoIP the same as SIP?
No. VoIP is the general idea of carrying voice over IP. SIP is one protocol, the most widely used one, that sets up and ends VoIP calls. The audio itself travels separately over RTP.
How much bandwidth does a VoIP call use?
With G.711 at 20 ms packets a call uses about 87 kbit/s each way including IP and Ethernet overhead. G.729 needs about 31 kbit/s each way.
Why does my VoIP call have one-way audio?
One-way audio is usually a NAT or firewall problem: signalling gets through but RTP packets are sent to a private or wrong address. SBCs, STUN/ICE or correct NAT settings fix it.