Bandwidth Requirements
Each VoIP call consumes bandwidth in both directions (upload and download). The exact amount depends on the codec used:
| Codec | Audio Bitrate | With IP Overhead | Quality |
|---|---|---|---|
| G.711 (PCMU/PCMA) | 64 kbps | 87 kbps | Excellent (toll quality) |
| G.722 | 64 kbps | 87 kbps | HD Voice (wideband) |
| G.729 | 8 kbps | 32 kbps | Good (compressed) |
| Opus | 6-128 kbps | Variable | Excellent (adaptive) |
Quick Formula: For G.711, multiply your peak concurrent calls by 100 kbps (with some overhead margin). So 20 simultaneous calls need approximately 2 Mbps of dedicated bandwidth in each direction.
Remember that bandwidth must be available in both directions. Many internet connections are asymmetric (faster download than upload), so your upload speed is typically the bottleneck. A 50/10 Mbps connection has enough upload bandwidth for about 100 G.711 calls, but only if no other traffic is competing for that upstream capacity.
Latency and Jitter
Latency is the time it takes for a voice packet to travel from sender to receiver. Jitter is the variation in latency between packets. Both critically affect call quality:
< 150ms
One-way latency target for high-quality calls. Conversations feel natural.
150-300ms
Noticeable delay. Speakers start talking over each other.
> 300ms
Unacceptable for voice. Conversations become very difficult.
Jitter should be kept below 30ms. Your PBX or phone uses a jitter buffer to smooth out packet arrival timing, but excessive jitter causes the buffer to overflow, resulting in choppy audio or dropped syllables.
Wi-Fi Warning: Wireless networks introduce significant and unpredictable jitter. For desk phones, always use wired Ethernet connections. Wi-Fi softphones should only be used on enterprise-grade wireless networks with proper QoS.
Packet Loss
Voice is much less tolerant of packet loss than data applications. While a web page loads fine with 2% packet loss, VoIP calls become noticeably degraded at much lower levels:
- 0-0.5% loss: No noticeable effect on call quality
- 0.5-1.5% loss: Minor artifacts - occasional clicks or very brief gaps
- 1.5-3% loss: Clearly degraded - missing words, robotic sound
- > 3% loss: Unusable for voice communication
Packet loss is usually caused by network congestion (too much traffic for available bandwidth), faulty cables or hardware, or overloaded switches/routers. The solution is almost always QoS - prioritizing voice packets over data traffic.
Quality of Service (QoS) Configuration
QoS ensures voice packets get priority treatment on your network, even when bandwidth is congested. This is the single most important network configuration for VoIP quality.
DSCP Markings
Voice packets should be marked with specific Differentiated Services Code Point (DSCP) values so routers and switches know to prioritize them:
! Define voice traffic class
class-map match-any VOICE
match dscp ef
match dscp cs3
! Define policy
policy-map QOS-POLICY
class VOICE
priority percent 30
set dscp ef
class class-default
fair-queue
! Apply to WAN interface
interface GigabitEthernet0/1
service-policy output QOS-POLICYTip: QoS only helps when there is congestion. If your internet connection is never saturated, QoS will not make a difference. But it is critical insurance for when traffic spikes do occur.
Network Design Best Practices
Dedicated Voice VLAN
Separate voice and data traffic on different VLANs. This isolates voice from data broadcast storms and makes QoS policies easier to apply.
Dedicated Internet Circuit
For high call volumes, consider a separate internet circuit for voice. This guarantees bandwidth and eliminates contention with data traffic.
PoE Switches
Use Power over Ethernet switches to power IP phones. This simplifies deployment and allows phones to share data ports via passthrough.
Redundant Paths
Dual internet connections with automatic failover protect against circuit outages. IPComms SIP trunks can register from multiple locations for redundancy.
Switch and Router Settings
Key settings to configure on your network equipment for optimal VoIP performance:
- LLDP-MED or CDP: Enable on switch ports connected to IP phones. This allows the switch to automatically assign phones to the voice VLAN and configure QoS.
- Spanning Tree PortFast: Enable on access ports to eliminate the 30-50 second delay phones experience when connecting to a port while STP converges.
- Storm Control: Configure broadcast storm control to prevent a data network problem from flooding the voice VLAN.
- MTU: Ensure consistent MTU (1500 bytes) along the entire path. Fragmented voice packets cause quality issues.
- NAT/Firewall: Configure SIP ALG (disable it if causing issues), open required ports for SIP (5060/5061) and RTP (typically 10000-20000).
SIP ALG: Many consumer and business routers have SIP Application Layer Gateway enabled by default. This feature often causes more problems than it solves, including one-way audio and registration failures. Try disabling SIP ALG if you experience issues.
Testing Your Network
Before deploying VoIP, test your network to ensure it meets the requirements:
# Test latency and packet loss to your SIP provider
ping -c 100 sip.ipcomms.net
# Check jitter with mtr (install: apt install mtr)
mtr --report --report-cycles 100 sip.ipcomms.net
# Test bandwidth with iperf3 (run server on remote end)
iperf3 -c your-server -u -b 1M -t 60
# Check for SIP ALG issues with ngrep
ngrep -W byline -d eth0 port 5060Target results for production VoIP deployment:
Ready to Deploy VoIP?
IPComms provides reliable SIP trunking with direct carrier interconnects for low-latency, high-quality calls. Our support team can help you validate your network readiness.