Choppy, robotic, or intermittently dropping audio on Twilio voice calls is caused by packet loss or jitter in the RTP media stream between the two call endpoints. Packet loss means RTP packets are being dropped somewhere in the network path, and the receiving endpoint's jitter buffer cannot reconstruct a smooth audio stream from the remaining packets. Jitter means RTP packets are arriving at irregular intervals (some arriving early, some late, some very late), overwhelming the jitter buffer and causing gaps in the reconstructed audio.
What Causes Choppy Audio
Network congestion on the path between the Twilio media server and your endpoint causes both packet loss and jitter: when a network link becomes saturated, routers drop packets from the tail of the queue, and RTP audio packets (which are small UDP datagrams) are particularly vulnerable to being dropped under congestion because they have no retransmission mechanism like TCP. Wi-Fi interference causes bursty packet loss that produces choppy audio specifically on calls from wireless devices: 2.4 GHz Wi-Fi is particularly susceptible to interference from Bluetooth devices, microwave ovens, and adjacent Wi-Fi networks, and the resulting loss pattern produces characteristic intermittent audio gaps rather than continuous choppiness. Insufficient jitter buffer configuration at the receiving endpoint causes choppiness even when packet loss is low: if the jitter buffer is too small to absorb natural network jitter, packets that arrive slightly late are discarded and treated as lost, producing audio gaps on a call that shows very low actual packet loss at the network level. CPU overload on the endpoint device or server running the SIP client causes the application to miss RTP packet processing deadlines, producing jitter in the outbound media stream from that endpoint: this is common on server-side PBX deployments that are handling more concurrent calls than the hardware can process efficiently.
How to Measure Packet Loss and Jitter
Open Twilio Call Insights for the affected call under Console, then Monitor, then Insights, then Voice and click on the specific Call SID: the Insights view shows MOS (Mean Opinion Score) for each direction of the call, packet loss percentage, and jitter in milliseconds, measured from RTCP data collected during the call. A MOS score below 3.5 indicates noticeable audio quality degradation; below 3.0 indicates poor quality that affects intelligibility; below 2.5 is very poor quality. Packet loss above 1 percent is perceptible as choppiness; above 3 percent produces significant audio gaps; above 10 percent makes speech nearly unintelligible. Jitter above 30 milliseconds is perceptible; above 50 milliseconds is problematic; above 150 milliseconds overwhelms typical jitter buffers and produces severe audio gaps. Run a network quality test from the same network as the affected endpoint using the Twilio Voice JavaScript SDK's Device.testPreflight() method or a standalone WebRTC network test tool: these tests measure packet loss and jitter between the client network and Twilio's media servers specifically, which is more relevant than a generic internet latency test.
How to Fix at the Network and Codec Level
For Wi-Fi-related packet loss, switch the affected devices to a 5 GHz Wi-Fi band (which is less congested than 2.4 GHz) or to a wired Ethernet connection: a wired connection eliminates wireless interference completely and typically reduces packet loss from the 1 to 3 percent range to under 0.1 percent. For network congestion, implement QoS (Quality of Service) rules on your network infrastructure that prioritize RTP UDP traffic (typically on ports 10,000 to 60,000) over other traffic types: on most enterprise routers and switches, this is configured by creating a DSCP (Differentiated Services Code Point) marking rule that marks RTP packets with the EF (Expedited Forwarding) DSCP value and configuring the network to give EF-marked traffic priority queuing. Switch from the G.711 codec (which uses 64 kbps of bandwidth per call) to the Opus codec (which uses 6 to 20 kbps and includes forward error correction that reconstructs dropped packets from redundant data in subsequent packets): if your PBX supports Opus, enabling it and setting it as the preferred codec in your SDP offer reduces the bandwidth consumed by each call and adds loss resilience through FEC. Increase the jitter buffer size on your SIP client or PBX: for Asterisk this is the jbenable=yes and jbmaxsize configuration in sip.conf; for 3CX this is in the audio settings; for the Twilio JavaScript Voice SDK this is the codecPreferences option, and a larger buffer absorbs more jitter at the cost of slightly increased latency.
When to Escalate
If Call Insights shows low packet loss and low jitter on the Twilio side but callers still report choppy audio, the packet loss is occurring between your PBX and the caller's PSTN handset rather than between your PBX and Twilio: this is a carrier-to-handset quality issue that requires investigation of the trunk connection between your PBX and the PSTN carrier, which is separate from the Twilio media path. Escalate to Twilio support when Call Insights shows high packet loss specifically on the Twilio-to-endpoint direction for calls that originate from multiple different client locations (not just one office network), as widespread directional loss suggests an issue with the Twilio media server cluster serving your region rather than a local network problem. A specialist is needed when choppy audio is intermittent and difficult to reproduce consistently, as intermittent audio quality issues require persistent monitoring with real-time RTCP data collection during failing calls, which requires custom instrumentation that goes beyond what the standard Call Insights dashboard provides.
Conclusion
Choppy Twilio audio is a packet loss or jitter problem that is measured through Call Insights RTCP data and fixed at the network layer (QoS, Wi-Fi to wired switch, congestion reduction) or codec layer (Opus with FEC, larger jitter buffer). If audio quality issues are affecting your production calls and you need help diagnosing the root cause, contact our team and we will analyze the Call Insights data and identify the fix within the hour.
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