How to configure Asterisk to accept and use Opus ptime=80 requested by endpoints?

Asterisk Asterisk SIP
1

Hi everyone,

I’m trying to force a packetization time (ptime) of 80ms for the Opus codec in my Asterisk setup. I have tried setting SDP_AUDIO_PTIME=80 in the dialplan like this:

exten = _X.,1,NoOp(Forcing ptime to 80)
same = n,Progress()
same = n,Set(SDP_AUDIO_PTIME=80)
same = n,Dial(PJSIP/${EXTEN})

However, when I check the SDP in the SIP INVITE, it still shows ptime=20, and the endpoints don’t seem to honor the 80ms packetization time.

Additional info:

  • Codec: Opus
  • direct_media is set to no
  • Dialplan reloads and PJSIP reloads done after changes
  • Tried setting Answer() before Dial, but no change
  • Endpoint is a softphone (or WebRTC client)
  • Tried opus.conf but unsure if it’s applied or configured correctly

I understand that Opus negotiates maxptime and ptime differently than legacy codecs, but I’d like to know:

  • What is the correct way to force Opus ptime=80 in Asterisk?
  • Is it possible at all to do it purely via dialplan or CLI?
  • Do I need to configure opus.conf or something else?
  • Are there any known limitations or workarounds to enforce this setting for Opus?
  • How can I verify the setting is actually applied during calls?

Any example configurations or tips would be highly appreciated.

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