Connect SIP peers. Manage calls and media.
Connect configured SIP peers, control calls and media, and manage operations from one web portal. TeleFinity SBC brings guided connectivity, routing, admission policies, optional media relay and searchable diagnostics to supported carrier and PBX integrations.
The controls between your SIP peers.
SIP network interconnection
Connect configured carrier, PBX and application peers through a back-to-back user agent (B2BUA) that manages separate call legs. Use UDP, TCP or TLS according to each peer’s requirements.
Guided connection setup
Create paired incoming and outgoing routes with the browser-based Connections wizard. Validate endpoint settings and review generated routes before saving.
DID and destination routing
Route calls using called numbers, exact matches and prefix rules. Distinct DID assignments help separate connections sharing a carrier source endpoint.
Target selection
Choose primary-target scanning or round-robin selection for new calls. Target capacity and temporary down-state influence eligibility; established calls are not migrated between servers.
Call admission and capacity
Configure shared connection limits, individual trunk and target concurrency, and calls-per-second limits. Control accepted work around the capacity planned for your environment.
Source and calling policies
Apply IP/CIDR source restrictions, caller identity rules, digest authentication, request-rate limits and international-call permissions. Combine policies with the network’s firewall configuration.
Signaling and media protection
Use SIP TLS with certificate validation and optional client certificates. Supported SDES-SRTP media can be terminated separately on each leg, subject to the required TLS and relay configuration.
Optional RTP media relay
Anchor media and rewrite supported SDP for each peer. Configure advertised public and private addresses to suit your network boundaries, alongside the required NAT and RTP/RTCP reachability.
Codec policies and announcements
Filter permitted audio and video codecs and configure WAV announcements using supported negotiated codecs. Codec filtering does not transcode live calls.
Live configuration changes
Apply supported ordinary route, codec and media-policy changes to new calls while existing calls retain their original configuration. Trunk revocation, listener maintenance and restarts need separate call-impact planning.
Live monitoring and SIP history
Search active calls, inspect both SIP legs and review captured SIP messages. Relayed calls expose per-leg RTP packet totals to help investigate asymmetric media paths.
Completed call records
Review completed CDRs with call identities, timing, final status, route and release details. Retain completed records for agreed service-review, investigation and archival workflows.
From connection requirements to validated calls.
01 — Define the connection
Identify carrier and application endpoints, transports, receiving numbers, calling permissions and expected call volumes.
02 — Configure and review
Use the connection wizard or supported trunk settings to define routes, admission limits, media behavior and peer policies. Validate and review before saving.
03 — Test both directions
Place and release real test calls, verify two-way media, and inspect Live monitoring, SIP history and CDR. Include hold/resume, cancellation, DTMF and policy checks relevant to the deployment.
Build the boundary your telephony environment needs.
Carrier-to-PBX connectivity
Place SBC between a SIP carrier and a PBX or telephony application to manage peer settings, routing, admission and diagnostics centrally.
Shared carrier infrastructure
Route distinct DID assignments from a shared carrier source to different application endpoints, with connection-specific capacity limits.
Public and private network boundaries
Use anchored media and per-network advertised addresses when peers sit on different networks. Plan firewall rules, NAT forwarding and reachable media ports.
Call troubleshooting
Follow a call through Live monitoring, SIP history and CDR using its inbound and outbound Call-IDs. Compare relay packet totals when media direction needs investigation.
Operate through one management portal.
Browser-based administration
Configure connections and trunks, review active calls, search SIP history and inspect completed records. Portal sign-in uses a password followed by a short-lived email verification code, with working SMTP required.
Configuration with call continuity
Validated supported live changes apply to new calls while established calls retain their settings. Configuration backups and revision checks support controlled changes; stop/restart and revocation actions have different call impacts.
Diagnostics in context
Use both call-leg identities, transport and route details to investigate issues. SIP history has configurable retention; protect diagnostic access and backups because message content can contain sensitive information.
Plan for your network, peers and operating model.
Windows and Linux deployment
Review Windows or Linux hosting options, protected portal access, persistent settings and record storage, certificate requirements and email delivery.
Network and media design
Confirm IPv4 addressing, listener ports, firewall and NAT rules, advertised media addresses and RTP/RTCP reachability. Qualify transport, codec and secure-media compatibility with the actual peers.
Capacity and operational readiness
Measure concurrency, call rate and resource use against your expected workload. Agree maintenance windows, storage retention, backups and recovery procedures. OPTIONS replies alone do not validate calls or media.
Operational readiness
Agree administrator responsibilities, diagnostic access, maintenance windows and a tested recovery approach. Validate the chosen configuration with actual calls in both directions.
Your SBC questions, answered.
Can each side use a different SIP transport?
Yes. Configure source and target transports independently, for example carrier UDP and application TCP, or supported TLS arrangements.
Will saving settings interrupt active calls?
Supported ordinary edits leave existing calls on their original configuration. Disabling or deleting a trunk ends its calls; allow-list changes can block active signaling. Stop, restart and listener maintenance affect calls and require planning.
Does codec filtering convert live audio?
No. Both peers need a permitted common codec. Announcement encoding is separate from live media transcoding.
Can browsers connect directly through WebRTC?
The supported media profile does not provide direct browser WebRTC connectivity using ICE/DTLS-SRTP. Discuss the connectivity required by your endpoints during solution planning.
Which secure media profile is supported?
SBC supports SDES-SRTP with AES_CM_128_HMAC_SHA1_80, requiring TLS signaling and media relay. The supported initial-offer profile covers one audio and optionally one video section. Confirm peer compatibility and RTCP handling during deployment.
Can portal users be limited to one customer?
All current portal accounts administer the whole instance. Shared carrier routing does not create separate portal tenants or read-only roles.
Do RTP packet counts measure voice quality?
They show accepted incoming and successfully sent outgoing RTP packets on each measured relay leg. They do not directly measure jitter, loss percentage or perceived speech quality. Direct-media calls are shown as not measured.
Does target fallback preserve calls during a server failure?
Target selection applies to new calls. The implemented scope does not provide replicated high availability or seamless recovery of established calls after a process or server failure.
Plan your SIP interconnection with TeleFinity.
Bring your carrier, PBX, routing, media and operational requirements. Explore the SBC configuration that fits your environment.