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Cybersecurity

VoIP SIP Header Analyzer, RTP Bandwidth & SS7 Risk Simulator (2026)

Parse and audit raw VoIP SIP INVITE/REGISTER signaling headers for Caller-ID spoofing, STIR/SHAKEN attestation levels (A/B/C), RTP codec bandwidth (G.711, Opus, G.729), and legacy SS7/Diameter telecom attack vectors.

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VoIP SIP Header Analyzer, RTP Bandwidth & SS7 Risk Simulator — Interactive Console
Runs locally in your browser • Instant output
From vs P-Asserted-Identity
From: +18009359935 | PAI: +37255591022
SPOOFED DISPLAY NUMBER!
STIR/SHAKEN PASSporT
NONE (Unverified / Unsigned Call)
PBX Scanner Fingerprint
ALERT: friendly-scanner/1.18 (sipvicious)
Per-Call Wire Rate: 87.2 kbps
Total SIP Trunk: 4.36 Mbps
Ready
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Blog / Documentation HTML Citation<a href="https://www.zerosuniverse.com/tools/sip-voip-ss7-telecom-attack-simulator/">VoIP SIP Header Analyzer, RTP Bandwidth & SS7 Risk Simulator — ZerosUniverse</a>

2026 Quick-Reference Cheat Sheet & Benchmark Table: VoIP SIP Header Analyzer, RTP Bandwidth & SS7 Risk Simulator

Quick Answer & 2026 Technical Summary (sip invite header analyzer voip calculator)Updated 2026 Standard

In baseline RFC 3261 SIP, the `From:` header (`From: "Bank Fraud Dept" <sip:18005550199@spoof.example>`) and `P-Asserted-Identity` (PAI) are plain-text fields. If an upstream wholesale SIP trunk does not enforce strict ANI validation or strip untrusted PAI headers, an attacker's PBX can assert any telephone number. Use this interactive sip invite header analyzer voip calculator above to test stir shaken identity header analyzer, voip rtp bandwidth codec calculator, and sip caller id spoofing detector locally in your browser with zero server uploads.

Target Keyword Spec: sip invite header analyzer voip calculator | Modules: Forensic SIP INVITE / REGISTER Header Parser • STIR/SHAKEN PASSporT (RFC 8224) Attestation Inspector • Precision VoIP RTP Codec & Trunk Bandwidth Calculator
Primary Focus: sip invite header analyzer voip calculator
Core Capability: stir shaken identity header analyzer
Privacy Mode: 100% Client-Side (Zero Upload)
Technical Parameter / ModuleStandard / Keyword SpecArchitecture & Validation RuleOperational Use Case (2026)
Forensic SIP INVITE / REGISTER Header Parserstir shaken identity header analyzerDissect Via hops, From/To display name vs. URI mismatches, P-Asserted-Ident...VoIP SOC Triage & Vishing Caller-ID Spoofing Forensics
STIR/SHAKEN PASSporT (RFC 8224) Attestation Inspectorvoip rtp bandwidth codec calculatorDecode base64url `Identity:` JWT headers and payloads to verify Full Attest...Enterprise SIP Trunk & SD-WAN QoS Capacity Planning
Precision VoIP RTP Codec & Trunk Bandwidth Calculatorsip caller id spoofing detectorCalculate exact Layer-2/Layer-3 Kbps and Mbps trunk capacity across G.711 (...Executive OPSEC & Telecom Threat Modeling
Execution & Privacy Architecture100% Client-Side WebCrypto / JS Sandbox0 Bytes Sent to External ServersSafe for internal SOC & authorized lab artifacts
NIST SP 800-53 / OWASP AlignmentOWASP ASVS v4.0.3 / NIST CSF 2.0Deterministic Rule & Header VerificationMaps findings to actionable hardening controls
Cryptographic & Entropy StandardSHA-256 / AES-256-GCM / Argon2id≥ 128-bit Effective Security MarginMeets 2026 post-quantum & zero-trust baselines
In-Depth ZerosUniverse Tutorial

What Are SS7 & VoIP Telecom Attacks?

Read our complete step-by-step editorial guide, architecture breakdown, and defensive best practices on ZerosUniverse.

Read Full Guide

How to Use VoIP SIP Header Analyzer, RTP Bandwidth & SS7 Risk Simulator

01

Paste a Raw SIP INVITE Packet or Load a Forensics Preset

Drop a SIP signaling capture into the analyzer—or load presets for Legitimate STIR/SHAKEN Attestation A, Vishing Spoofed PAI, or SIPVicious Toll-Fraud Recon.

02

Audit Spoofing Indicators & STIR/SHAKEN PASSporT Claims

Review the parsed Via path, From vs. P-Asserted-Identity alignment, decoded STIR/SHAKEN `attest` grade (A/B/C), and SDP media encryption (`RTP/AVP` cleartext vs `RTP/SAVP` SRTP).

03

Calculate Concurrent VoIP RTP Trunk Bandwidth & PPS

Switch to the RTP Calculator tab, choose your codec (G.711, Opus, G.729), packet interval (20ms = 50 pps), and concurrent channels to compute total Mbps.

04

Explore the SS7 MAP / Diameter Roaming Attack Matrix

Inspect the interactive SS7 call-flow simulator to see how GSMA FS.11 Category 1/2/3 Home Routing and SMS firewall rules block rogue HLR/VLR queries.

Key Capabilities & Technical Architecture

Forensic SIP INVITE / REGISTER Header Parser

Dissect Via hops, From/To display name vs. URI mismatches, P-Asserted-Identity (PAI), Contact routing, User-Agent scanner signatures (sipvicious/friendly-scanner), and SDP media attributes.

STIR/SHAKEN PASSporT (RFC 8224) Attestation Inspector

Decode base64url `Identity:` JWT headers and payloads to verify Full Attestation (A), Partial Attestation (B), Gateway Attestation (C), and origination tracking (`origid`).

Precision VoIP RTP Codec & Trunk Bandwidth Calculator

Calculate exact Layer-2/Layer-3 Kbps and Mbps trunk capacity across G.711 (PCMU/PCMA), G.729, Opus, and G.722 codecs with 10ms/20ms/30ms packetization, VLAN 802.1Q, and SRTP/WireGuard overhead.

SS7 MAP / Diameter Telecom Attack & Defense Simulator

Explore how legacy SS7 MAP (`SendRoutingInfoForSM`, `ProvideSubscriberInfo`, `UpdateLocation`) and 4G Diameter messages enable SMS interception and cell-tower tracking—and how SS7 firewalls block them.

Practical Use Cases

VoIP SOC Triage & Vishing Caller-ID Spoofing Forensics

Paste raw SIP INVITE packet captures from Asterisk, FreeSWITCH, or Kamailio SBCs to detect spoofed `From` headers, missing STIR/SHAKEN `Identity` tokens, and rogue SIP scanners.

Enterprise SIP Trunk & SD-WAN QoS Capacity Planning

Compute exact DSCP EF (Expedited Forwarding) bandwidth reservations and packets-per-second (PPS) load for 50 to 5,000 concurrent calls with SRTP encryption.

Executive OPSEC & Telecom Threat Modeling

Evaluate how SS7 `UL` (UpdateLocation) and `SRI-SM` roaming attacks compromise SMS 2FA codes and why high-risk executives must migrate to FIDO2 hardware keys.

Frequently Asked Questions (FAQs)

How do attackers spoof Caller ID in VoIP SIP calls?+

In baseline RFC 3261 SIP, the `From:` header (`From: "Bank Fraud Dept" <sip:18005550199@spoof.example>`) and `P-Asserted-Identity` (PAI) are plain-text fields. If an upstream wholesale SIP trunk does not enforce strict ANI validation or strip untrusted PAI headers, an attacker's PBX can assert any telephone number.

How does STIR/SHAKEN (RFC 8224 / RFC 8588) authenticate caller identity?+

The originating carrier signs a JSON Web Token called a PASSporT inside the SIP `Identity:` header using its STI-CA X.509 private key. The payload binds the calling number (`orig`), called number (`dest`), timestamp (`iat`), and Attestation Level: 'A' (Full: carrier knows the customer and their right to use the number), 'B' (Partial: customer known, number unverified), or 'C' (Gateway: international/legacy transit dump).

Why does a 64 kbps G.711 voice call actually consume 87.2 kbps on the network?+

At standard 20ms packetization (50 packets per second), each packet carries 160 bytes of G.711 audio payload plus 12 bytes of RTP header, 8 bytes of UDP header, 20 bytes of IPv4 header, and 18 bytes of Ethernet L2 framing (58 bytes of header overhead per packet). Multiply 218 total bytes × 8 bits × 50 pps = 87.2 kbps per call.

How does an SS7 MAP `SendRoutingInfoForSM` attack intercept SMS 2FA codes?+

Signaling System 7 (SS7) was built in the 1970s on implicit trust between global telecom operators. An attacker with leased access to an SS7 global title sends a fake `UpdateLocation` (UL) message to the victim's Home Location Register (HLR), claiming the victim is roaming on the attacker's MSC/VLR node. Subsequent `MT-ForwardSM` SMS messages are routed straight to the attacker.

How do you prevent VoIP RTP audio eavesdropping on internal networks?+

Standard SDP media (`m=audio ... RTP/AVP`) transmits voice streams in cleartext UDP, allowing anyone with a packet sniffer (Wireshark) to replay calls with one click. Enforce TLS 1.3 for SIP signaling (SIPS on port 5061) and SRTP (`RTP/SAVP` with SDES or DTLS-SRTP) for AES-encrypted audio transport.