Audio Delay and Talking Over Each Other

When Your Phones Sound Bad: Part 7 of 12

The conversation feels wrong. You say something, and there's a noticeable pause before the other person responds. Or worse, you both start talking at the same time because neither of you can tell when the other has finished speaking. The audio isn't choppy, it isn't cutting out, and the call is clear. It's just delayed, like watching a video that's slightly out of sync.

This is a latency problem, and while it's one of the simpler VoIP issues to understand conceptually, it can be one of the harder ones to fix because the cause often lives somewhere outside your direct control.

What latency means for a phone call

Latency is the time it takes for data to travel from one point to another. On a VoIP call, latency is the delay between the moment you speak and the moment the other person hears it. Some amount of latency is unavoidable. The audio has to be captured by the microphone, encoded into digital data, packetized, transmitted across the network, received, decoded, and played through the speaker. Even under ideal conditions, this takes a few milliseconds.

The question is how much latency before it becomes a problem. The general thresholds are:

Under 150 milliseconds one way: the conversation feels natural. Most people don't notice any delay at all.

150 to 300 milliseconds one way: the delay is noticeable. Conversations still work but feel slightly unnatural. You might occasionally talk over each other.

Over 300 milliseconds one way: the delay is significant and makes fluid conversation difficult. Both parties frequently talk over each other, and there are long uncomfortable pauses as each person waits to make sure the other has finished speaking.

For reference, a phone call between two VoIP phones on the same local network might have 5 to 20 milliseconds of latency. A call between two offices in the same city over a good internet connection might have 20 to 50 milliseconds. A call from the US to Europe might have 80 to 120 milliseconds just from the physical distance the data has to travel. A call over a geostationary satellite connection has 600 or more milliseconds, as discussed in the Getting the Internet Right series.

Where latency comes from

Latency is cumulative. Every component in the path between your mouth and the other person's ear adds some delay. The total latency is the sum of all these contributions.

Encoding delay. The codec that encodes your voice into digital data takes time to process. G.711 has minimal encoding delay because it's a simple encoding. G.729 has slightly more because it uses compression. Opus can vary depending on its configuration. The encoding delay is typically 5 to 40 milliseconds and is not usually the primary culprit.

Packetization delay. Audio is collected into packets covering a fixed time interval (usually 20 milliseconds). The phone has to wait for a full packet's worth of audio before it can send anything. This adds 20 milliseconds of delay by definition.

Network delay. The time for the packet to travel from your phone to the other person's phone. This includes the time to traverse your local network, your internet connection, the internet backbone (potentially crossing multiple networks), the other party's internet connection, and their local network. Network delay varies based on geographic distance, the number of routing hops, congestion levels, and the type of connections involved.

Jitter buffer delay. The receiving phone uses a jitter buffer to smooth out variations in packet arrival times. The buffer holds packets for a short period to allow late packets to arrive before playing them in sequence. A larger jitter buffer adds more delay but handles jitter better. A smaller buffer adds less delay but is more sensitive to timing variations. The jitter buffer typically adds 20 to 80 milliseconds.

Processing delay. The PBX, SBC, or any other intermediary that processes the call adds a small amount of delay. If the PBX is transcoding (converting the audio from one codec to another), the transcoding process adds additional delay, potentially 20 to 40 milliseconds per transcoding step.

Diagnosing latency

The first step is measuring it. Running our VoIP quality test will show you the latency your connection adds under realistic voice traffic conditions. For a quick check, the simplest approach is a ping test to a relevant destination. Ping your VoIP provider's SIP server or a point as close to the other end of the call as possible. The round trip time (RTT) in the ping response gives you the network portion of the latency. Divide the RTT by two for the approximate one way network delay.

If the ping RTT to your VoIP provider is 40 milliseconds, that's about 20 milliseconds one way network delay. Add the encoding, packetization, and jitter buffer delays (collectively maybe 40 to 60 milliseconds on each end), and the total one way latency is roughly 100 to 120 milliseconds. That's fine. If the ping RTT is 200 milliseconds, the total one way latency is probably around 160 to 180 milliseconds, which is getting into the range where the delay becomes noticeable.

If you have access to your PBX or VoIP provider's call detail records, many systems report RTP statistics including round trip time for each call. This gives you a direct measurement rather than the approximation from a ping test.

Common causes of excessive latency

Geographic distance. Data travels through fiber at roughly 200,000 km/s (about two-thirds the speed of light in vacuum), but over thousands of miles that still adds up. A call from New York to London (approximately 5,570 km great-circle distance) adds approximately 35 to 45 milliseconds each way from propagation delay alone, with real-world routing adding some overhead on top of that. You can't fix physics. If you're making a lot of international calls and latency is an issue, the only mitigation is ensuring everything else in the chain is as fast as possible so you're not adding avoidable delay on top of the unavoidable distance.

Too many network hops. Every router between you and the destination adds a small amount of processing delay. If your traffic is taking an inefficient path through the internet (routing through multiple ISPs, going through a distant exchange point rather than a local one), the accumulated delay can be significant. A traceroute to your VoIP provider's server shows you the path and the delay at each hop. If you see a large jump in latency at a specific hop, that hop is the bottleneck.

ISP congestion. When your ISP's network is congested, packets wait in queues at congested points, which adds delay. This is the same congestion that causes choppy audio, but at lower levels of congestion the effect is increased latency rather than packet loss. Latency that's worse during peak hours (midday, early evening) and better during off peak hours is a classic congestion pattern.

VPN routing. If your VoIP traffic is routed through a VPN, the VPN adds delay in several ways. The traffic has to be encrypted and encapsulated, which takes processing time. The traffic may travel to a VPN concentrator that's geographically distant from the direct path, adding physical distance. And the VPN endpoint may be on a congested network. If you're experiencing latency on calls and using a VPN, test with the VPN disconnected to see if the latency drops.

Transcoding. If your PBX is converting between codecs (for example, your phone uses Opus internally and the SIP trunk uses G.711), each transcoding step adds delay. If the call is being transcoded on both the inbound and outbound legs, the delay adds up. Minimizing transcoding by using the same codec end to end, or at least ensuring only one transcoding step occurs, reduces latency.

Oversized jitter buffer. If the jitter buffer on the phone is set to a large fixed value, it adds unnecessary delay even when network conditions are good. Most modern phones use an adaptive jitter buffer that adjusts its size based on actual network conditions. If your phone allows you to configure the jitter buffer, set it to adaptive (or dynamic) rather than a large fixed value.

What you can control and what you can't

Some sources of latency are within your control: your local network, your QoS configuration, your codec selection, your jitter buffer settings, your VPN routing, and whether your PBX is transcoding. These are all things you can optimize.

Some sources are partially within your control: your ISP (you can choose a provider with a better network, or upgrade to a connection with lower latency), your VoIP provider (some providers have better network peering than others, which affects latency to different destinations).

Some sources are outside your control: geographic distance, the routing decisions of upstream ISPs, and congestion on networks you don't operate. For these, the best you can do is ensure that everything within your control is optimized so you're not adding avoidable delay to the unavoidable baseline.

If latency is consistently high on calls to specific destinations, it's worth asking your VoIP provider whether they have a point of presence closer to those destinations, or whether their peering with the relevant carriers could be improved. Good VoIP providers invest in network peering specifically to reduce latency on voice traffic.

Next up: Phones Not Ringing for Incoming Calls, covering what to check when the call arrives but the phone stays silent.