Voice Stream (RTP)

The Real-time Transport Protocol (RTP) is the network protocol used to carry audio and video data over IP networks during a live communication session. In a VoIP phone call, the actual voice audio — encoded as a continuous stream of data packets — is transmitted via RTP between the caller's phone, the carrier network, the enterprise telephony infrastructure, and the AI Voice Gateway. RTP operates alongside SIP: SIP handles call signalling (setup and teardown), while RTP carries the media (the actual audio). For voice AI systems, the quality and latency of the RTP audio stream directly affects ASR accuracy and the perceived naturalness of the interaction.

For enterprise teams, Voice Stream (RTP) matters because real-world outcomes depend on how the capability is integrated, governed, and measured — not just on the underlying technology. For voice AI systems, the quality and latency of the RTP audio stream directly affects ASR accuracy and the perceived naturalness of the interaction. 

Key Points

• Network protocol carrying live audio and video data over IP networks in real time
• Works alongside SIP: SIP handles signalling; RTP carries the actual voice media
• Audio quality and RTP latency directly affect ASR accuracy and conversation naturalness
• Packet loss, jitter, and delay in RTP streams must be minimised for voice AI quality
• NiCE Cognigy Voice Gateway optimises RTP media handling for enterprise voice AI