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Version: Next — unreleased

SIP signalling and telephony edge

The SIP edge that carriers connect to, the methods, transports, timers and responses it supports, how it hands calls to the voice service and how SIP traffic is captured for diagnosis. Browse the feature map or configuration finder.

Availability: Next. Entries that depend on an optional feature, a role or a deployment choice name it in their own Availability line.

Features in this family​

SIP edge proxy​

A SIP proxy at the platform boundary receives every carrier call, identifies the trunk, applies edge policy and forwards the call to a voice host.

For: Integrators, Deployment operators

LevelWhere to change itOwner section
Services and containersTelPro environment(DOMAIN_TELPRO, SIP_DOMAIN, KAMAILIO_INSTANCE_ID, KAMAILIO_LOG_LEVEL)TelPro variables

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Related features: SIP transports and ports, Voice host selection by the edge, Public and private edge addresses (NAT).

SIP transports and ports​

The edge listens for SIP over UDP, TCP and TLS on published ports and uses a separate internal port toward the voice hosts.

For: Integrators, Deployment operators

LevelWhere to change itOwner section
Services and containersTelPro environment(SIP_PORT, SIPS_PORT, SIP_TLS_PORT, INTERNAL_SIP_PORT, SIP_PORT_INTERNAL, TLS_ENABLED)TelPro variables

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Related features: SIP over TLS at the edge, Network ports, allowed sources and firewall, SIP edge proxy.

Supported SIP methods​

The edge accepts the SIP requests needed for calls, transfers and keepalives and answers other methods with a defined rejection.

For: Integrators

LevelWhere to change itOwner section
Services and containersSIP edge routing (no customer setting)(INVITE, ACK, BYE, CANCEL, OPTIONS, REFER, NOTIFY, UPDATE, PRACK, INFO)SIP methods

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Related features: Responses generated by the edge, OPTIONS keepalive, Reliable provisional responses (100rel and PRACK).

Responses generated by the edge​

Some requests are answered by the edge itself, for example keepalives, unknown trunks or blocked admission, so carriers can tell an edge decision from a voice service answer.

For: Integrators, Deployment operators

LevelWhere to change itOwner section
Services and containersSIP edge routing (no customer setting)(sl_send_reply)Responses generated by the edge

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Related features: Supported SIP methods, Number lookup and routing responses, Rejected and unknown SIP traffic.

OPTIONS keepalive​

The edge answers SIP OPTIONS pings from carriers so their trunk health checks see the platform as reachable.

For: Integrators

LevelWhere to change itOwner section
Services and containersSIP edge routing (no customer setting)(OPTIONS)Request handling and trunk health

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Related features: Supported SIP methods, Trunk health probes.

Reliable provisional responses (100rel and PRACK)​

Provisional responses can be sent reliably and acknowledged with PRACK when a carrier requires it, with the mode chosen per voice deployment.

For: Integrators, Deployment operators

LevelWhere to change itOwner section
Services and containersTelSys environment(SIP_100REL_MODE)Voice variables

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Related features: Supported SIP methods.

SIP transaction and dialog timers​

Fixed edge timers limit how long a request waits for an answer, how long a call may ring and how long an established dialog is kept.

For: Integrators

LevelWhere to change itOwner section
Services and containersSIP edge configuration (no customer setting)(fr_timer, fr_inv_timer)Timers and retransmission

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Related features: Maximum call duration, Supported SIP methods.

SDP, codecs and number format at the edge​

The edge and voice hosts negotiate a defined codec set and SDP shape and expect numbers in a stated format, which carriers need to match during onboarding.

For: Integrators

LevelWhere to change itOwner section
Services and containersVoice and edge media configuration (no customer setting)Codecs and DTMF

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Related features: Voice codecs on SIP calls, Public and private edge addresses (NAT), Supported SIP methods.

Public and private edge addresses (NAT)​

The edge advertises its public address in SIP and SDP while using private addresses inside the platform, so media flows correctly across NAT.

For: Deployment operators, Integrators

LevelWhere to change itOwner section
Services and containersTelPro environment(PUBLIC_IP, PRIVATE_IP, EXTERNAL_IP, HOST_LAN_IP)TelPro variables

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Related features: SIP edge proxy, Media floating address.

SIP termination headers​

When the platform ends or transfers a call it adds a Reason header with a cause and short text, plus a transfer reason on transfer INVITEs, so the carrier side can log why.

For: Integrators

LevelWhere to change itOwner section
Services and containersVoice runtime (format fixed; limits per app)(Reason, X-Transfer-Reason)Diagnostic header formats

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Related features: Platform termination causes and SIP mapping, Hang up from a flow or a bot, Final SIP response for calls that never connect, Transfer reason on the new call leg, Optional SIP header budget.

Voice host selection by the edge​

The edge sends each new call to a healthy voice host chosen by a balancing method and skips hosts whose heartbeat is missing or that are draining.

For: Deployment operators

LevelWhere to change itOwner section
Platform settingsAdmin Settings › Server Groups › Load Balancing Method(lbMethod)Configure a server group
Services and containersTelPro environment(TELSYS_LB_METHOD)TelPro variables

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Related features: Load balancing across voice instances, Instance heartbeats and usage reports, Draining an edge.

Voice service control interface (ARI and AMI)​

The voice runtime controls calls on the telephony hosts through their REST and manager interfaces, with session limits that cap how many calls one host can carry.

For: Deployment operators

LevelWhere to change itOwner section
Services and containersTelSys and TelPhi environment(ARI_URL, ARI_USER, ARI_PASSWORD, ARIAPP, ARIENABLED, ARIBINDMASK, ARIHTTPSESSIONLIMIT, AMI_HOST, AMI_PORT, AMI_USER, AMI_SECRET)Voice variables

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Related features: Voice service operation, Voice host selection by the edge, Concurrent call limit per voice instance and group, Voice service and media diagnostic logs.

Handoff of an inbound call to the voice runtime​

After the edge forwards an INVITE, the telephony host passes the channel to the voice runtime, which looks up the number and starts the flow.

For: Deployment operators, Integrators

LevelWhere to change itOwner section
Services and containersVoice runtime (no customer setting)(StasisStart)Follow the request path

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Related features: Voice service control interface (ARI and AMI), Number lookup and routing responses.

Rejected and unknown SIP traffic​

Requests from unknown sources or matching no trunk are rejected and tagged in the logs, so scanning traffic is visible without reaching a voice host.

For: Deployment operators

LevelWhere to change itOwner section
Services and containersSIP edge routing (no customer setting)Rejected calls and security tagging

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Related features: Responses generated by the edge, Trunk identification and allowed IPs, SIP edge log records, Edge source verification and call token checks.

SIP edge health​

Each edge reports whether it is up and accepting calls, which operators check before and after changes.

For: Deployment operators

LevelWhere to change itOwner section
Services and containersTelPro health endpointCheck edge health

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Related features: Draining an edge, Edge instance overview.