carriers-sip
RTP
Real-time Transport Protocol, the stream of small packets that carries the actual voice audio of a VoIP call between the two endpoints.
RTP stands for Real-time Transport Protocol. It is the stream of small packets that carries the actual voice of a call — the words you hear — between the two ends of a VoIP connection. While SIP (Session Initiation Protocol) does the negotiating to set up and end a call, RTP is the part that ships the audio once everyone is connected.
It helps to picture a phone call as two jobs. SIP is the operator who places the call and rings the line. RTP is the wire the voice rides on after they connect. These travel on different network ports, which is why a call can sometimes ring and connect fine but have no sound — the signaling worked while the audio path did not.
The voice inside each RTP packet is compressed by a Codec, so the codec choice and the RTP stream go hand in hand. Because RTP is real-time, the packets need to arrive quickly and in order. When they get jumbled in timing you hear Jitter, and when some never show up you hear Packet loss — both turn smooth speech into a robotic or choppy mess.
For troubleshooting, RTP is where most audio complaints live. If a caller says "I can hear you but you can't hear me," the RTP stream is flowing one direction only — a classic One-way audio symptom, usually caused by a firewall or router blocking the audio ports. Knowing that RTP is separate from the signaling is the first step to fixing these problems instead of blaming the dialer.
One detail trips up a lot of newcomers: RTP uses a wide range of network ports, not a single fixed one, and routers that use address translation often need help passing those ports through correctly. This is the whole reason NAT traversal exists — to make sure the audio finds its way to a machine sitting behind a home or office router. When an agent can connect and hear ringing but the call goes silent the instant it answers, the audio path is almost always the culprit, and the fix lives in the firewall and router settings rather than anywhere in VICIdial. Because the protocol is so sensitive to timing, RTP also rewards a steady, low-delay connection; a link that is fast on average but bursty can still produce ugly audio if the packets do not arrive at an even pace.
Related terms
Codec
The method that compresses and decompresses voice audio for a VoIP call, trading off between sound quality and how much network bandwidth each call uses.
Jitter
Jitter is the uneven arrival timing of voice packets on a call, which can make audio sound choppy or robotic even when little data is actually lost.
One-way audio
One-way audio is when one person on a call can be heard but cannot hear the other, almost always a problem with how the audio stream is routed.
Packet loss
Packet loss happens when voice data packets fail to reach their destination, leaving gaps that sound like clipped words or robotic, broken audio.
SIP (Session Initiation Protocol)
The standard signaling protocol that sets up, manages, and ends internet phone calls — how VICIdial talks to phones and carriers.
VoIP
Voice over Internet Protocol, the technology that sends phone calls as data packets over the internet instead of over traditional copper phone lines.