Optimizing SIP ALG Bypass Strategies for Genesys Cloud Direct Inward Dialing (DID) Number Porting

How-To-Articles
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Optimizing SIP ALG Bypass Strategies for Genesys Cloud Direct Inward Dialing (DID) Number Porting

What This Guide Covers

This guide configures and validates SIP Application Layer Gateway (ALG) bypass mechanisms across carrier Session Border Controllers, enterprise firewalls, and Genesys Cloud SIP trunk endpoints to ensure pristine signaling during DID number porting. You will implement transport-level isolation, Session Description Protocol (SDP) preservation rules, and validation workflows that prevent call setup failures and media path corruption during number migration.

Prerequisites, Roles & Licensing

  • Licensing Tier: Genesys Cloud CX 2 or CX 3 (SIP Trunking requires CX 2 minimum; WEM add-on required if you integrate porting validation with workforce management dashboards)
  • Granular Permissions: Telephony > Trunk > Edit, Telephony > Number > Edit, Telephony > Call > Read, Architect > Routing > Edit
  • OAuth Scopes: telephony:trunk:write, telephony:number:read, telephony:call:read, architect:routing:read
  • External Dependencies: Administrative access to carrier SBC or SIP interconnect provider, enterprise firewall/SBC management console (Palo Alto, Fortinet, Cisco ASA, or equivalent), packet capture tooling (Wireshark, tcpdump, or Genesys Cloud SIP trace export)

The Implementation Deep-Dive

1. Perimeter Firewall & NAT SIP ALG Neutralization

SIP ALG functions at Layer 7 to inspect and modify SIP signaling. During DID porting, carriers route inbound traffic through temporary prefixes or modified Request-URI patterns. An active SIP ALG intercepts these modifications, rewrites the Contact header, and alters SDP connection addresses. Genesys Cloud edge infrastructure expects the original signaling payload to maintain routing integrity. If the ALG rewrites the SDP, media streams fail to establish because the Genesys media server responds to an IP address that the ALG has already invalidated.

We disable SIP ALG at the perimeter by isolating SIP traffic on dedicated TCP ports and enforcing connection tracking rules that bypass deep packet inspection. UDP 5060 remains vulnerable to ALG interception on most commercial firewalls. We shift SIP signaling to TCP 5060 or TCP 5061 (TLS) and configure the firewall to treat these ports as stateful passthrough tunnels without payload inspection.

Configuration Approach:
On Linux-based firewalls or host-level network appliances, we implement nftables or iptables rules that explicitly exclude SIP ports from ALG modules. On commercial enterprise firewalls, we disable the SIP application filter entirely for the traffic selector matching the Genesys Cloud SIP Trunk IP ranges.

# nftables example: Bypass SIP ALG for Genesys Cloud IP ranges
table inet fw4 {
    chain input {
        # Block ALG inspection for SIP signaling ports
        tcp dport { 5060, 5061, 5004-5020 } ct state established,related accept
        # Explicitly drop ALG module hooks if loaded
        meta nfproto ipv4 tcp dport 5060 accept
        meta nfproto ipv6 tcp dport 5060 accept
    }
}

For commercial firewalls, we configure a dedicated security policy that matches the Genesys Cloud SIP Trunk destination IPs and sets the Application Override to SIP-TCP with Inspection: Disabled. We also pin the source NAT translation to preserve the original Caller ID and P-Asserted-Identity headers. NAT reflection must be disabled for these rules. NAT reflection causes the firewall to answer internal SIP OPTIONS probes with its own public IP, breaking Genesys health checks during porting validation.

The Trap: Leaving SIP ALG enabled on a firewall profile that claims to only inspect “malformed” SIP traffic. Most ALG implementations trigger on standard SIP INVITE messages containing SDP bodies. During porting, carriers often send provisional 100 Trying or 180 Ringing responses with temporary SDP attributes. The ALG intercepts these, rewrites the c= line, and injects a Connection: close header. Genesys Cloud interprets this as a signaling timeout and returns a 408 Request Timeout. The porting batch fails silently for 30-40% of numbers until the ALG cache flushes.

Architectural Reasoning: We prioritize TCP over UDP for SIP during porting because TCP provides sequence numbering and retransmission logic that survives carrier network congestion. Porting events generate burst signaling as carriers validate number portability requests. UDP drops under congestion cause immediate INVITE failures. TCP ensures the complete SIP dialog reaches the Genesys edge before we evaluate routing. We also eliminate UDP fragmentation issues that occur when carriers embed large X-Carrier-Number or P-Asserted-Identity headers in SIP messages.

2. Carrier SBC & Genesys Cloud SIP Trunk Transport Alignment

The carrier SBC sits between the public switched telephone network and Genesys Cloud. During DID porting, the carrier SBC maps ported numbers to the Genesys SIP Trunk IP. We must align the SBC transport profile with Genesys Cloud expectations to prevent double NAT translation and SDP mangling.

We configure the Genesys Cloud SIP Trunk via the API to enforce TCP transport, disable SDP rewriting, and set media handling to passthrough. Passthrough media handling ensures Genesys Cloud forwards the original SDP payload without modifying codec negotiation or connection addresses. This is critical during porting because carriers may test numbers using different codec preferences before final cutover.

API Configuration Payload:
We use the Genesys Cloud v2 API to provision the SIP Trunk with precise transport and media constraints. The following POST request creates a trunk optimized for ALG-free porting.

POST https://api.mypurecloud.com/api/v2/telephony/providers/edge/trunks
Authorization: Bearer <OAUTH_TOKEN>
Content-Type: application/json

{
  "name": "CarrierPortingTrunk-Primary",
  "description": "SIP Trunk for DID porting with ALG bypass enforcement",
  "transportType": "TCP",
  "encryptionType": "NONE",
  "mediaHandling": "PASSTHROUGH",
  "sipUri": "sip:gen-edge-12345.mypurecloud.com",
  "fromDomain": "carrier-sbc.example.com",
  "toDomain": "gen-edge-12345.mypurecloud.com",
  "ipAddresses": [
    "203.0.113.10",
    "203.0.113.11"
  ],
  "numberListIds": [
    "d8a7c9e1-4f2b-4c5d-8e9a-1b2c3d4e5f6a"
  ],
  "authentication": {
    "username": "porting_trunk_user",
    "password": "secure_trunk_credential_hash",
    "realm": "gen-edge-12345.mypurecloud.com"
  },
  "options": {
    "sendPai": true,
    "sendPidf": false,
    "allowSdpRewrite": false,
    "natTraversal": false
  }
}

We set allowSdpRewrite to false and natTraversal to false. The carrier SBC handles NAT boundaries. Genesys Cloud should never attempt to fix SDP addresses when the SBC has already translated them. We also enable sendPai to preserve P-Asserted-Identity headers. These headers carry the original ported DID number. Genesys Cloud routing conditions rely on P-Asserted-Identity to assign calls to the correct queue or skill group.

On the carrier SBC side, we configure a SIP profile that disables SDP modification and forces TCP transport. We also disable SIP normalization features that rewrite the Request-URI to match local numbering plans. Ported numbers often retain their original geographic prefix. Normalization breaks Genesys routing patterns that match against the full E.164 string.

The Trap: Enabling natTraversal on the Genesys SIP Trunk while the carrier SBC operates in a fully NAT-aware mode. The Genesys edge adds a Via header with a public IP address and modifies the SDP c= line to point to the SBC public IP. The SBC then rewrites it again to its own public IP. This creates a recursive NAT loop. Inbound calls from ported DIDs hit the SBC, which forwards to Genesys, but Genesys sends media to the SBC public IP. The SBC drops the RTP stream because it does not match an established signaling session. You experience one-way audio or immediate disconnects on 100% of ported numbers.

Architectural Reasoning: We separate NAT responsibilities between the SBC and Genesys Cloud. The SBC owns the public-to-private translation boundary. Genesys Cloud operates in a controlled cloud environment and expects pristine signaling. By disabling natTraversal and allowSdpRewrite, we force Genesys to trust the SBC translation. This reduces CPU overhead on the Genesys edge, lowers latency during porting validation, and prevents media path mismatches. We also isolate porting traffic on a dedicated trunk to prevent porting signaling bursts from consuming resources on production trunks.

3. Porting Workflow Integration & SIP Header Preservation

DID porting introduces temporary SIP headers that carriers use to route calls before the final Local Routing Guide (LRG) update propagates across the PSTN. Headers such as X-Carrier-Number, X-Porting-ID, and modified P-Asserted-Identity values carry routing intelligence. SIP ALG implementations often strip custom headers or normalize them to standard RFC 3261 formats. We must preserve these headers through the firewall, SBC, and Genesys routing layer.

We configure the carrier SBC to pass custom headers without modification. We also create a Genesys Cloud routing pattern that evaluates P-Asserted-Identity and Request-URI fields to route ported calls to a validation queue. This queue allows your team to verify call quality, codec negotiation, and DTMF pass-through before promoting numbers to production queues.

SBC Header Preservation Configuration (Generic SIP Profile):

sip-profile porting-trunk-profile
  disable-sdp-modification
  preserve-headers X-Carrier-Number X-Porting-ID P-Asserted-Identity
  transport tcp
  max-dialogs 5000
  retransmit-timeout 3s
  invite-timer 30s

Genesys Cloud Routing Condition:
In Genesys Architect, we create a routing condition that matches inbound calls from the porting trunk. The condition evaluates the P-Asserted-Identity field against a dynamic list of ported numbers. We route matches to a dedicated validation queue.

Condition: P-Asserted-Identity contains "ported_number_list_dynamic"
Action: Route to Queue "Porting_Validation_Queue"
Fallback: Route to Queue "Default_Inbound_Queue"

We also configure the firewall to whitelist these custom headers in the SIP inspection bypass policy. Some firewalls enforce strict RFC compliance and drop SIP messages with non-standard headers when ALG is partially enabled. We override this by setting the SIP policy to Header Preservation: Strict Pass-Through.

The Trap: Allowing the carrier SBC to rewrite P-Asserted-Identity to a normalized local number. During porting, carriers often normalize the header to match their internal numbering plan. Genesys Cloud routing conditions expect the full E.164 DID string. When the SBC rewrites the header, the routing condition fails to match. Calls route to the fallback queue, which may lack the correct skill-based routing or speech analytics tags. You experience misrouted calls, compliance recording failures, and WEM skill mismatch alerts.

Architectural Reasoning: We treat SIP headers as immutable routing tokens during the porting window. The P-Asserted-Identity header carries the legally registered DID number. Modifying it breaks regulatory compliance logging and Genesys Cloud’s built-in number validation. We preserve headers end-to-end to ensure routing accuracy, compliance recording alignment, and accurate WEM reporting. We also isolate porting traffic on a dedicated trunk to prevent header stripping policies from affecting production traffic. This isolation simplifies troubleshooting and allows us to apply aggressive header preservation rules without impacting legacy SIP trunks.

Validation, Edge Cases & Troubleshooting

Edge Case 1: Carrier-Imposed SDP Contact Rewriting During Partial Ports

The Failure Condition: Inbound calls from ported DIDs successfully ring and connect, but RTP audio fails to establish. Genesys Cloud SIP traces show 200 OK responses, but PCAP captures reveal mismatched SDP c= addresses. The call disconnects after 3-5 seconds with a 488 Not Acceptable Here or silent RTP timeout.

The Root Cause: The carrier SBC rewrites the SDP Contact header to its own public IP address during partial porting windows. The perimeter firewall SIP ALG, even in bypass mode, intercepts the TCP SIP stream and appends a Connection: keep-alive header that forces the SBC to re-evaluate the SDP. Genesys Cloud receives a modified SDP and responds with RTP to the SBC public IP. The SBC drops the media because it does not match the original signaling session.

The Solution: Disable SDP modification entirely on the carrier SBC profile for the porting trunk. Configure the firewall to drop any SIP messages containing Connection: keep-alive or Connection: close headers on port 5060/5061. Force Genesys Cloud to use mediaHandling: PASSTHROUGH and allowSdpRewrite: false. Verify SDP integrity using a Genesys Cloud SIP trace export. Compare the original c= line from the INVITE with the SDP in the 200 OK response. They must match exactly. If they differ, adjust the SBC SIP profile to enforce preserve-sdp-contact: true.

Edge Case 2: TLS Session Resumption Failure Under High Porting Volume

The Failure Condition: During bulk DID porting (500+ numbers), SIP INVITE success rate drops to 60-70%. Genesys Cloud returns 503 Service Unavailable errors. Firewall logs show TLS handshake timeouts and session ticket drops. The issue resolves after the porting batch completes.

The Root Cause: The perimeter firewall SIP ALG emulation module intercepts TCP 5061 TLS traffic and attempts to decrypt session resumption tickets. During high-volume porting, the ALG cache overflows and drops TLS tickets. Genesys Cloud retries handshakes, consuming SBC resources. The SBC returns 503 errors because it cannot maintain dialog state under retransmission storms.

The Solution: Pin TLS Server Name Indication (SNI) to the Genesys Cloud SIP Trunk FQDN. Configure the firewall to bypass TLS inspection for TCP 5061 traffic matching the Genesys IP ranges. Increase the firewall connection tracking timeout for SIP TCP sessions to 300 seconds. On the carrier SBC, enable TCP keepalive probing to maintain session state during TLS retries. Verify TLS session resumption using openssl s_client -connect <gen-edges-ip>:5061 -tls1_2. Confirm that the SBC and firewall both report Session-ID: <stable-hash> across multiple porting INVITE attempts. If session IDs fluctuate, disable TLS ALG emulation entirely for the SIP traffic selector.

Official References