Network Requirements for VoIP — QoS & Bandwidth
Overview
For a voice call to sound clear and natural, three network conditions must be met: packets must arrive quickly, arrive consistently, and not be dropped. When any of these degrade beyond acceptable limits, callers hear echo, robotic audio, clipping, or complete audio loss.
This document defines the QoS and bandwidth requirements your network must meet to deliver high-quality calls on the Exotel platform.
QoS Requirements
These are the network performance thresholds Exotel measures and recommends for all voice traffic.
Metric | Recommended | Acceptable | What happens if exceeded |
|---|---|---|---|
One-way latency | ≤ 130 ms | ≤ 150 ms | Echo, conversational overlap |
Round-trip time (RTT) | ≤ 260 ms | ≤ 300 ms | Unnatural conversation delay |
Jitter | ≤ 20 ms | ≤ 40 ms | Choppy, robotic audio |
Packet loss | 0% | ≤ 1% | Clipping, gaps in speech |
MOS score (G.711) | ≥ 4.2 | ≥ 3.6 | Poor call experience |
MOS score (OPUS) | ≥ 4.3 | ≥ 3.6 | Poor call experience |
Exotel's platform contributes < 20 ms of processing delay. The remaining latency budget is your network, ISP, and endpoint.
Why these thresholds? Latency and RTT are derived from ITU-T G.114, which specifies ≤ 150 ms one-way for high-quality voice. Jitter and packet loss targets follow ITU-T Y.1541. MOS is measured using the ITU-T G.107 E-model — the real-world ceiling for G.711 is ~4.4 and OPUS is ~4.5, so the recommended targets reflect consistently good quality, not just a passing grade.
Bandwidth Requirements
Per Call
Codec | Used For | Bandwidth per Call |
|---|---|---|
G.711 | SIP Trunking, IP Calling, PSTN | 100 kbps |
OPUS | WebRTC, Audio Streaming, Voice for AI | 60–80 kbps |
Always provision 1.5× your peak concurrent call bandwidth to allow headroom for signaling and traffic bursts.
Concurrent Call Capacity (G.711)
Concurrent Calls | Minimum Bandwidth | Recommended Link |
|---|---|---|
10 | 1 Mbps | 1.5 Mbps |
50 | 5 Mbps | 7.5 Mbps |
100 | 10 Mbps | 15 Mbps |
500 | 50 Mbps | 75 Mbps |
Applicability by Channel
Not all QoS parameters apply the same way across channels. Use this as a quick reference.
Parameter | SIP Trunking | IP Calling | WebRTC | Audio Streaming | PSTN |
|---|---|---|---|---|---|
Latency / RTT | Applicable | Applicable | Applicable | RTT affects AI response | Carrier |
Jitter | Applicable | Applicable | Applicable | TCP absorbs jitter | Carrier |
Packet loss | Applicable | Applicable | Applicable | TCP retransmits | Carrier |
Bandwidth planning | Applicable | Applicable | Applicable | Per stream | Carrier |
Notes:
- PSTN — QoS applies only to the IP leg between your network and Exotel's media gateway. The PSTN leg is carrier-managed.
- Audio Streaming — Uses TCP/WebSocket. Jitter and packet loss behave differently from RTP. What matters most is round-trip latency for AI pipeline responsiveness (target RTT ≤ 300 ms).
- WebRTC — DSCP marking is typically stripped by browsers. QoS must be applied at the network edge.
Network Checklist Before Go-Live
- Average ping to Exotel SIP endpoints < 50 ms; no spikes > 150 ms
- UDP jitter test (iperf3) for ≥ 5 minutes under load — target < 20 ms
- Packet loss test under simulated call load — must be < 1%
- Bandwidth confirmed for peak concurrent call volume (see table above)
- SIP ALG disabled on all routers and firewalls
- UDP ports 10000–20000 open bidirectionally
- 10 test calls placed and MOS reviewed in Exotel dashboard — target ≥ 4.0
Common Issues
Symptom | Likely Cause | Fix |
|---|---|---|
One-way audio | SIP ALG enabled | Disable SIP ALG on router/firewall |
Choppy / robotic voice | Jitter > 40 ms or packet loss > 1% | Check network congestion; enable DSCP EF (46) marking |
Calls drop at 30–60 s | Firewall UDP timeout too low | Set UDP session timeout ≥ 180 s |
Echo | Latency > 150 ms | Use the nearest Exotel PoP; enable echo cancellation |
DTMF not detected | RTP UDP ports blocked | Verify UDP 10000–40000 fully open |
SIP 429 errors | CPS/CPM limit exceeded | Reduce call pace; contact support to increase limits |
SIP ALG is the #1 cause of VoIP issues. Always disable it — on Cisco ASA, Fortinet, Juniper, Mikrotik, pfSense, and ISP-provided routers.
Exotel Platform Commitments
What Exotel guarantees | Value |
|---|---|
Platform processing latency | < 20 ms |
Inter-datacenter latency | < 10 ms |
Media server uptime | 99.5% SLA |
Redundancy | Multiple PoPs, automatic failover |
Encryption | SIP-TLS (signaling) + SRTP (media) |
Quality reporting | MOS scores in CDR and dashboard (RTCP-XR) |
Exotel guarantees QoS within its own infrastructure only. Your last-mile ISP, LAN/WAN, and endpoint are outside Exotel's control.
For support: [email protected] · docs.exotel.com