Kamailio Dispatcher for AI Voice Scaling in 2026: Round-Robin Is Not Enough

A round-robin SIP dispatcher works for the first hundred concurrent AI calls. By call number 200 you discover that one bridge node has 80% of the GPU pressure and another has 10%. Kamailio 6.0's dispatcher does better, but only if you configure it past defaults.

Background

flowchart LR
 Phone["PSTN caller"] --> Carrier["Carrier"]
 Carrier -- "SIP INVITE" --> SBC["Session Border Controller"]
 SBC -- "SIP" --> PBX["Twilio / Asterisk"]
 PBX -- "RTP · Opus" --> Bridge["AI Voice Gateway"]
 Bridge --> AI["OpenAI Realtime"]
 AI --> Bridge
 Bridge --> PBX

Kamailio is a SIP server, not a media gateway; it routes signaling. The dispatcher module distributes SIP requests across a pool of downstream destinations using one of several algorithms: round-robin, hash over From/To/Call-ID, weight-based, call-load-based, and a few others. Kamailio 6.0 (released 2025) and 6.1 (2026) refined the call-load algorithm and added new health-check options.

For AI voice in 2026, Kamailio fronts a fleet of bridge nodes (FreeSWITCH or your own FastAPI bridge) that each terminate calls and connect to OpenAI Realtime. The dispatcher picks which bridge gets the next call. Round-robin is the obvious default; under real production load with non-uniform call durations and tool-call-heavy conversations, it leaves bridges unbalanced. Call-load dispatching tracks active calls per destination and routes to the least-loaded node.

Technical deep-dive

# /etc/kamailio/dispatcher.list
1 sip:bridge-1.callsphere.ai:5060;weight=10
1 sip:bridge-2.callsphere.ai:5060;weight=10
1 sip:bridge-3.callsphere.ai:5060;weight=8
1 sip:bridge-4.callsphere.ai:5060;weight=10

# kamailio.cfg snippet
loadmodule "dispatcher.so"
modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list")
modparam("dispatcher", "ds_ping_interval", 10)
modparam("dispatcher", "ds_probing_threshold", 3)
modparam("dispatcher", "ds_probing_mode", 1)

route {
 if (is_method("INVITE")) {
 # Algorithm 9 = call-load (active calls per destination)
 if (!ds_select_dst("1", "9")) {
 # 8 = weight-based fallback
 ds_select_dst("1", "8");
 }
 t_on_failure("FAIL_FALLBACK");
 route(RELAY);
 }
}

failure_route[FAIL_FALLBACK] {
 if (t_check_status("5[0-9][0-9]")) {
 ds_mark_dst("ip"); # mark this destination as inactive
 ds_next_dst(); # try next
 t_relay();
 }
}

The call-load algorithm requires Kamailio to track active dialogs (mod_dialog) so it knows which node currently has how many calls. The weight-based algorithm uses static weights from the dispatcher list, which works if your nodes are heterogeneous (different vCPU sizes).

For AI voice the additional knob is "tool-call density". A bridge node handling a Healthcare AI call with frequent EHR lookups consumes more CPU than one handling a Salon AI call that only books appointments. Pure call count under-counts that. A custom XHTTP route can let bridges report load back to Kamailio and feed a custom_dispatcher state table.

CallSphere implementation

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Implementation steps

1. Build a fleet of bridge nodes that each register with Kamailio (or are statically listed in dispatcher.list).
2. Set ds_ping_interval to 5-15 seconds with TCP/TLS pings; mark unhealthy nodes inactive within a minute.
3. Use algorithm 9 (call-load) primary for AI voice; calls vary in duration too much for hash-based to balance.
4. Set per-destination weights matching vCPU + GPU capacity; reset weights when scaling up.
5. Enable failure_route to retry on 5xx responses against the next destination automatically.
6. Configure mod_dialog so dispatcher can track active dialog counts.
7. Expose Kamailio dispatcher metrics via xhttp and scrape into Prometheus.
8. Pre-fetch capacity: at 90% bridge utilization across the fleet, scale up the auto-scaling group; do not wait until 100%.

FAQ

Can Kamailio handle media too? No. Kamailio is signaling only. Media goes to FreeSWITCH, RTPProxy, or directly between endpoints (depending on topology).

Does Twilio expose this kind of dispatch internally? Twilio's edge does its own load balancing across their bridges. You do not see or tune it.

What about Kubernetes-native SIP? Kamailio runs in Kubernetes fine; the harder part is keeping SIP UDP and TCP/TLS ports addressable through the LoadBalancer service. Use a NodePort or hostPort for SIP.

How many concurrent calls per Kamailio node? A single well-tuned Kamailio handles tens of thousands of CPS and millions of concurrent dialogs. The bottleneck is almost always the bridges.

What is new in Kamailio 6.0/6.1? 6.0 stabilized HTTP/2 client; 6.1 (2026) added improved dispatcher metrics and finer tlsf integration. KamailioWorld 2026 highlighted production deployments mixing dispatcher with WebSocket-backed mobile push.

Sources

• Kamailio: DISPATCHER Module documentation
• Sinologic: Kamailio as a Load Balancer for Asterisk - Practical Guide 2026
• Sinologic: KamailioWorld 2026 Future of VoIP

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