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 hello\@exotel com · docs exotel com