VoIP quality problems on shared networks — jitter, packet loss, choppy audio — are almost always fixed by QoS. Quality of Service prioritizes voice packets over data so calls get through cleanly even when the network is busy.
VoIP is extremely sensitive to network conditions. RTP packets must arrive within tight timing windows — jitter above 40ms causes audio degradation, and packet loss above 1% causes choppy calls. On a shared network without QoS, a single large file transfer or video stream can starve voice traffic and destroy call quality.
QoS solves this by marking voice packets with a high-priority DSCP value and configuring routers to serve those packets first, before data traffic. Even on a congested network, voice gets through.
Have a capture of the failure? Upload it to the online PCAP analyzer — SIP and RTP are extracted automatically and findings come back severity-ranked, free tier included. Or continue below for the manual approach.
DSCP (Differentiated Services Code Point) is a 6-bit field in the IP header that marks packet priority. For VoIP:
| Traffic Type | DSCP Class | DSCP Value | Binary |
|---|---|---|---|
| RTP media (voice/video) | EF (Expedited Forwarding) | 46 | 101110 |
| SIP signaling | CS3 or AF31 | 24 or 26 | 011000 |
| General data | BE (Best Effort) | 0 | 000000 |
LLQ gives a strict priority queue for EF traffic. Voice packets are always served before data. The voice queue has a bandwidth guarantee and a maximum to prevent it from starving data entirely.
Putting voice on a dedicated VLAN is the most reliable QoS approach — it physically separates voice traffic from data at the switch level, not just at the router.
WAN links are where QoS matters most — they're typically the bottleneck. Key principles:
For SD-WAN deployments, most platforms have voice-aware QoS policies built in. Configure the voice application category to use EF marking and strict priority delivery.
The key test: run a speed test while on a VoIP call. With QoS properly configured, the call should remain clear even at 100% bandwidth utilization. Without QoS, the call will degrade or drop.
Use DSCP EF (Expedited Forwarding, value 46) for RTP media packets and DSCP CS3 (value 24) or AF31 (value 26) for SIP signaling. DSCP EF ensures voice packets get strict priority treatment in router queues. In Asterisk set tos_audio=ef in asterisk.conf. In Linux use iptables mangle to mark UDP packets in the RTP port range with DSCP 46.
A G.711 (PCMU/PCMA) VoIP call uses approximately 87 kbps per call including IP, UDP, and RTP headers. G.729 uses approximately 32 kbps per call. Opus at full quality uses around 510 kbps but can be configured as low as 6 kbps. For QoS planning, calculate concurrent calls times per-call bandwidth and reserve that plus 20% headroom on your WAN link.
DSCP operates at Layer 3 (IP header) and is preserved end-to-end across routed networks. 802.1p CoS operates at Layer 2 (Ethernet frame) and only applies within a switched VLAN segment — it is stripped at router boundaries. For VoIP, use both: 802.1p CoS=5 for within the LAN/VLAN, and DSCP EF=46 for WAN and routed segments.
Upload your PCAP to SIPSymposium. The analyzer extracts RTCP reports, measures jitter and packet loss, and identifies whether the issue is network congestion that QoS would fix.