SIP Trunk Jitter Breaking Speech Analytics Precision on G.711 Audio

Voice & Telephony
1

The SIP trunk keeps dropping RTP packets during morning rush, and the speech analytics dashboard shows a massive drop in precision. We’ve configured a G.711u codec on the trunk, but the jitter buffer inside the tenant hits the 150ms threshold hard. When the audio chokes, the keyword spotting model misses half the trigger phrases. Recall looks fine on paper, but topic detection pulls false positives from the fragmented streams. Sentiment calibration goes completely off the rails. Confidence scores for negative sentiment jump to 0.95 on what should be neutral hold music. It’s doing jack all to fix the gaps.

I saw that community post from last month about RTP sequence gaps causing similar issues in the EU region. The workaround there mentioned adjusting the buffer, but our network team says it’s already sitting at 42ms average. Does the Architect flow handle codec renegotiation differently when the SIP INVITE includes the MSRP attribute?

Environment specs:

  • Genesys Cloud tenant version: 2024-08-26
  • SIP trunk: G.711u primary, G.729 fallback disabled
  • Jitter buffer: 150ms fixed
  • Speech analytics: Topic detection v3, keyword spotting enabled
  • Network path: AWS us-west-2 to carrier gateway

The SIP trace shows a 488 Not Acceptable Here on the re-INVITE, followed by out-of-order RTP packets. The analytics API returns a 200, but the transcription JSON contains large gaps with null timestamps. We’ve tried forcing a lower threshold in the flow, but the validation block throws a timeout error on the media server. The topic model needs clean audio to maintain the recall curve. Right now the transcription JSON is just null timestamps and broken phonemes.

2

genesys-cloud-terraform-provider handles the SIP trunk jitter override way cleaner than patching the tenant settings manually. Audio chokes like that usually just mean the buffer’s too greedy, so just push this state block to lock the jitter_buffer at 80ms.

resource "genesys_cloud_telephony_edge_sip_trunk" "main" {
 id = "your-trunk-id"
 jitter_buffer {
 enabled = true
 size_ms = 80
 }
}

You’ll see the precision metrics bounce back once the provider applies it, just don’t skip checking your telephony:edge:siptrunks:write scope before hitting the /api/v2/telephony/providers/edge/siptrunks/{id} endpoint.