---
title: "PJSIP Configuration & Codec Transcoding Secrets — WhatsMCP Blog"
description: "Optimizing pjsip.conf parameters for flawless G.711u to Opus conversion under high-throughput concurrent call loads."
url: "https://whatsmcp.com/blog/pjsip-configuration-and-codec-transcoding"
---

Optimizing pjsip.conf parameters for flawless G.711u to Opus conversion under high-throughput concurrent call loads.

## Codec Negotiation Order

List the codec your PBX actually terminates first — `allow=g711u` before `allow=opus` on the PBX-facing endpoint — so the negotiated media leg never round-trips through a third format you didn't ask for:

```ini
[pbx-facing]
type=endpoint
transport=transport-udp
context=pbx-inbound
disallow=all
allow=g711u
allow=opus
dtmf_mode=rfc4733
```

## Concurrency Tuning

Transcoding is CPU-bound, not bandwidth-bound, so the parameter that actually caps concurrent call capacity is the transcoding worker's own stream limit, not anything in `pjsip.conf` itself. Every call that mixes G.711u and Opus on the same leg holds one transcode session for its full duration.

## The One Setting Everyone Forgets

`direct_media=no` has to be set on both endpoints of a transcoded bridge, not just one. The asymmetric case is the one that passes testing with a single call and then drops audio only once the media path has to route through the transcoder under load.