VoIP Bandwidth Per Call by Codec
VoIP audio is compressed using codecs before being transmitted over the network. Different codecs offer different trade-offs between audio quality and bandwidth consumption. The codec in use is typically negotiated automatically between your VoIP phone and your provider's SIP server.
| Codec | Bandwidth Per Call | Audio Quality |
|---|---|---|
| G.711 (ulaw/alaw) | 87 Kbps | Standard quality — telephone grade |
| G.729 | 31 Kbps | Good quality with compression — most efficient |
| G.722 (HD Voice / Wideband) | 87 Kbps | High definition, significantly clearer than G.711 |
| Opus (WebRTC-based VoIP) | 6–510 Kbps adaptive | Excellent quality; adapts to network conditions |
| Video VoIP (basic SD) | 500 Kbps | 360p or 480p video with voice |
| Video VoIP (HD) | 1.5–3 Mbps | 720p or 1080p video conference quality |
Planning for Simultaneous Calls
For businesses or households with multiple concurrent VoIP lines, calculate total required bandwidth by multiplying the per-call bandwidth by the maximum number of simultaneous calls. Always provision more bandwidth than your peak estimate to leave headroom for other internet traffic and protocol overhead.
| Simultaneous Calls | Bandwidth Required | Practical Note |
|---|---|---|
| 1–5 calls | ~1 Mbps | Any broadband plan is sufficient |
| 6–15 calls | 2–5 Mbps | Bandwidth fine; prioritize QoS setup |
| 16–30 calls | 5–10 Mbps | Dedicated internet or fiber recommended |
| 30+ calls | 10–25 Mbps | Business fiber with SLA; enterprise QoS required |
Note: Latency and jitter are the real constraints at all call volumes. A 1 Mbps connection with jitter under 20ms and zero packet loss outperforms a 100 Mbps connection with 50ms jitter for VoIP quality.
VoIP Quality Thresholds: Latency, Jitter, and Packet Loss
The ITU-T G.114 standard defines the technical thresholds for acceptable VoIP quality. These metrics are measured on the network connection, not the VoIP device itself. A speed test that measures jitter and packet loss will show you whether your connection meets these thresholds.
| Metric | Acceptable | Degraded | Unacceptable |
|---|---|---|---|
| One-way latency | <150ms (ITU standard) | 150–200ms | >200ms — noticeable delay in conversation |
| Jitter | <30ms | 30–50ms | >50ms — choppy, robotic audio |
| Packet loss | <1% | 1–3% | >3% — severe degradation, missing words |
Why QoS Is Critical for VoIP
QoS (Quality of Service) is a router-level feature that assigns priority levels to different types of network packets. Without QoS, your router treats all packets equally. When a household member begins a large file download or starts streaming 4K video, the router's outbound queue fills with data packets, causing brief but regular delays in VoIP packet transmission.
Those delays manifest as jitter. Even a 10-second download spike that causes 80ms of jitter is enough to make a VoIP call momentarily unintelligible. QoS resolves this by marking VoIP packets as high priority and processing them before lower-priority bulk data traffic — ensuring that even a saturated connection delivers VoIP packets on time.
Setting Up QoS for VoIP
Most business-grade and prosumer routers support QoS configuration. The specific menus vary by manufacturer, but the principles are consistent:
DSCP Marking
VoIP traffic should be marked with DSCP EF (Expedited Forwarding, value 46) for voice RTP streams. SIP signaling traffic should be marked DSCP CS3. Most business routers from Cisco, Netgear Business, and Ubiquiti UniFi support DSCP marking rules. Ubiquiti UniFi's "VoIP" traffic profile applies these markings automatically.
Port-Based Prioritization
If your router does not support DSCP, prioritize VoIP by port number instead. The key ports to prioritize are:
- UDP 5060 — SIP signaling (call setup and teardown)
- UDP 5061 — SIP over TLS (encrypted signaling)
- UDP 16384–32767 — RTP media streams (the actual voice audio)
Bandwidth Reservation
Some routers allow you to reserve a guaranteed minimum bandwidth for VoIP traffic. Reserve at least 500 Kbps per expected concurrent call, which provides comfortable headroom above the actual codec requirement.
Popular VoIP Provider Bandwidth Recommendations
| VoIP Provider | Minimum Bandwidth Per Call | Notes |
|---|---|---|
| RingCentral | 100 Kbps | Recommends QoS and wired connections |
| Vonage Business | 90 Kbps | Uses G.711 and G.729 codecs |
| Nextiva | 100 Kbps | Recommends dedicated VLAN for VoIP |
| 8x8 | 100 Kbps | Publishes network readiness assessment tool |
| Dialpad | 100 Kbps | WebRTC-based; uses Opus codec adaptively |
Frequently Asked Questions
How much bandwidth does VoIP use?
A standard VoIP call using the G.711 codec uses approximately 87 Kbps per concurrent call including overhead. For planning, budget 100 Kbps per simultaneous call. Ten concurrent calls consume roughly 1 Mbps — a tiny fraction of any modern broadband connection. Bandwidth is almost never the limiting factor for VoIP quality.
Why does my VoIP call sound choppy?
Choppy audio in VoIP calls is caused by jitter or packet loss, not slow internet speed. Jitter means packets are arriving at uneven intervals, causing the audio decoder's buffer to empty and produce gaps. Packet loss means some audio frames are missing entirely, causing clicks and dropped syllables. Run a speed test to check your jitter value — if it exceeds 30ms or your packet loss exceeds 1%, those metrics are the cause of poor audio quality, not your download or upload speed.
What internet speed do I need for 10 VoIP lines?
Ten simultaneous VoIP calls require approximately 1 Mbps of dedicated bandwidth. Any standard broadband plan easily meets this threshold. The actual requirement is connection quality: jitter must stay below 30ms and packet loss below 1% even when other devices are downloading or streaming. Configuring QoS on your router to prioritize VoIP traffic is the single most impactful step for a 10-line VoIP deployment.
Does VoIP work on a 10 Mbps connection?
Yes, a 10 Mbps connection has approximately ten times more bandwidth than VoIP needs. Even 30 simultaneous calls consume only 3 Mbps. The concern with a 10 Mbps connection is not capacity but congestion: if other applications saturate the connection and cause jitter, VoIP quality degrades. QoS configuration on your router ensures VoIP packets are prioritized and delivered consistently even when the connection is under load.