Operations on Ulai Docs

Deployment

Mon, 01 Jan 0001 00:00:00 +0000

The gateway is distributed as a container image. A deployment is one container on a host with a routable public IP, run with host networking.

docker run

docker run -d --name ulai-sip \
 -e ULAI_CONTROL_PLANE_URL=https://stgcp.ulai.co.in \
 -e SIP_ROUTING_REDIS_URL='redis://USER:PASSWORD@redis.ulai.co.in:6379' \
 -e SIP_PUBLIC_IP=148.113.58.51 \
 -e SIP_LISTEN_ADDR=0.0.0.0:5060 \
 -e SIP_TRANSPORT=udp \
 -e SIP_RTP_HOST=0.0.0.0 \
 -e SIP_RTP_PORT_LOW=10000 \
 -e SIP_RTP_PORT_HIGH=10500 \
 -e SIP_HTTP_PORT=8082 \
 --network host \
 asia-south1-docker.pkg.dev/arctic-operand-415316/ulai/sip:v6

Three things in that command are load-bearing:

--network host — see networking. Publishing the RTP range with -p spawns hundreds of userland proxies and rewrites the INVITE’s source address, which the trunk ACL then rejects with 403.
SIP_PUBLIC_IP — the address the carrier is told to send media to. It must be the host’s own routable address, and never 0.0.0.0.
SIP_RTP_HOST=0.0.0.0 — what the RTP sockets bind to. On a cloud VM the public IP usually is not on any interface, so binding to it fails outright.

Pass secrets with --env-file rather than -e if the host’s process list is readable:

Networking

Mon, 01 Jan 0001 00:00:00 +0000

Ports

Port	Protocol	Purpose
`8082`	TCP	HTTP API — `/health`, `/calls`, `/sip/originate`
`5060`	UDP and/or TCP	SIP signalling (`5061` conventionally for TLS)
`10000–10500`	UDP	RTP media, one even port per call

The RTP range sizes the gateway’s concurrency: roughly 250 simultaneous calls at the default range, since allocation uses even ports only by RTCP convention. Widen SIP_RTP_PORT_LOW/HIGH to carry more.

The carrier must be able to reach SIP_PUBLIC_IP on the SIP port and the whole RTP range. A firewall that allows 5060 but not 10000–10500 produces a call that connects and then has no audio — the single most common misconfiguration.

Observability

Mon, 01 Jan 0001 00:00:00 +0000

Reading the log

Every line of a call is prefixed with the number and the leg it arrived on, so a busy gateway’s log can be read one call at a time:

[+919363399639] [sip-originate] call answered — call_id=3e114ad4…, joining room 527f78a3…
[+919363399639] [sfugw] webrtc state: connected
[+919363399639] [sfugw] DTMF: 5
[+919363399639] [sip-originate] call finished (session 527f78a3…, call_id 3e114ad4…)

Prefix	Emitted by
`[sip-gw]`	The inbound path
`[sip-originate]`	The outbound path
`[sfugw]`	The room/WebRTC bridge
`[control-plane]`	The control-plane wire log
`[runtime]`	Startup runtime tuning

grep on the number gives one call; grep on the sip_call_id joins the gateway’s log to carrier CDRs and to the published telecom events.

Troubleshooting

Mon, 01 Jan 0001 00:00:00 +0000

The gateway will not start

sip-sfu-gateway: ULAI_CONTROL_PLANE_URL is required (e.g. https://stgcp.ulai.co.in)
SIP_PUBLIC_IP must be a routable address, not 0.0.0.0 — it is advertised to the carrier

Configuration is validated once, and every problem is reported together — fix them all, then restart. 0.0.0.0 in SIP_PUBLIC_IP is rejected on purpose: it is not a listen address, it is what the carrier is told to stream media to.

Other startup failures: