stuck on a weird audio latency issue affecting our apac agents using the genesys cloud webrtc softphone. we have 15 byoc trunks configured across singapore and tokyo, primarily using bandwidth and local telcos. the issue is not consistent across all trunks but spikes significantly during peak hours (0900-1100 sgt) on the primary bandwidth trunk.
the softphone connects fine via udp, and the initial sip registration is successful. however, agents report a 200-400ms one-way audio delay that makes conversations difficult. i have verified the network path and packet loss is under 0.1%. the real-time analytics for the webrtc connections show stable jitter values (<10ms) at the client level, but the purview analytics for the trunk calls show high jitter (>50ms) during the same windows.
i suspect the issue lies in the rtp payload negotiation or the carrier’s handling of packetization intervals. the bandwidth trunk is configured with a fixed payload size of 20ms, but i noticed some sip trace logs showing the carrier responding with varying packetization intervals in the sdp answer, which might be causing buffer bloat on the softphone side.
has anyone seen this specific discrepancy between client-side webrtc metrics and trunk-side purview metrics? i am looking for best practices on forcing specific rtp codecs or payload sizes in the outbound routing configuration to mitigate this. also, should i be adjusting the webrtc softphone settings in the admin console to prefer specific codecs like opus over g.711 for better jitter tolerance? any insights on how to align the sdp negotiation to prevent the carrier from dictating suboptimal packetization would be greatly appreciated. currently testing with a small group of agents in singapore before rolling out any config changes.
The simplest way to resolve this is to stop looking at the softphone client and check your SBC logs for packet loss, as that is outside my recording export scope.
SIP 408 Request Timeout
I handle chain of custody for digital channels, not voice trunks.
check genesyscloud_edge_byoc_trunk config. ensure media_region aligns with agent location. mismatch causes extra hops.
resource "genesyscloud_edge_byoc_trunk" "main" {
media_region = "ap-southeast-1"
}
verify jitter buffer in SBC. terraform handles registration, not rtp path.
I typically get around this by checking the media region alignment first, because Zendesk doesn’t have the same concept of regional media servers for voice. In Zendesk, everything is just “the cloud,” but in Genesys Cloud, if your BYOC trunk is pointing to a media region that is far from your agents or your SBC, you get that extra hop latency. The suggestion above about media_region is spot on.
Since we are migrating from Zendesk, it is easy to forget that voice traffic needs to be routed as locally as possible. For our APAC agents in Singapore and Tokyo, ensure the media_region in your BYOC trunk configuration matches the edge location where the SBC is terminating. If the SBC is in Singapore (ap-southeast-1) but the trunk is pointing to a US region, the audio will travel all the way back to the US and then to the agent, causing that 200-400ms delay.
Here is the corrected Terraform payload to ensure the media region is explicitly set to the local edge. This avoids the default fallback which might not be optimal for your specific trunk setup.
{
"trunk_id": "your-byoc-trunk-id",
"media_region": "ap-southeast-1",
"edge_id": "singapore-edge-id",
"description": "Primary Bandwidth Trunk - APAC Local Media"
}
Also, check the jitter buffer settings on the SBC. In Zendesk, there was no manual jitter buffer tuning for voice, so this is a new variable to manage. If the jitter buffer is too high, it adds latency. If it is too low, you get choppy audio. Start with the Genesys recommended default and adjust based on the SBC logs during peak hours. This usually resolves the latency spike without needing to change the trunk configuration itself.