[Freeswitch-users] Mod_Opus :Add fmtp maxplaybackrate=8000 in OPUS codec offer.Version 1.6.8-15-99de0ad~64bit

Jonathan Hunter hunterj91 at hotmail.com
Wed Jul 12 14:33:02 UTC 2017


Hi Dragos,


Thank you for the fast reply!


You are correct in terms of call flow.


I am originating from a WebRTC client on Janus, which then sends a SIP invite into FreeSWITCH , I then reply to that to set it to 8K, narrowband, this works perfectly ! 😊


However I then bridge the call out to a carrier, and I want to offer in the SDP the maxplaybackrate=8000, however I cant seem to do this, and this is with and without late-negotiation enabled.


Am I missing something or do you have any debug suggestions for me please?


Many  thanks


Jon

---------- Forwarded message ----------
From: Oancea, Dragos <dragos.oancea at vonage.com<mailto:dragos.oancea at vonage.com>>
Date: Wed, Jul 12, 2017 at 1:14 PM
Subject: Re: [Freeswitch-users] Mod_Opus :Add fmtp maxplaybackrate=8000 in OPUS codec offer.Version 1.6.8-15-99de0ad~64bit
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>


Hi Jon,

If both sides follow the RFC , then you should be able to get narrowband end to end.
How do you originate the call ?
Sounds like you originate a call from a device and then you bridge to the callee, but you have inbound-late-negotiation on, so the OPUS config XML is not loaded, hence fmtp cannot be changed according to config.

Regards,
Dragos


On Wed, Jul 12, 2017 at 1:04 PM, Jonathan Hunter <hunterj91 at hotmail.com<mailto:hunterj91 at hotmail.com>> wrote:

Hi Guys,

We are currently testing OPUS and Im looking to lock down things to a maxplaybackrate of 8000HZ.

This works fine if FreeSWITCH is the receiver (Offer Answer), however if I then send an invite out from FreeSWITCH (bridge the same call)it doesnt seem to add the  fmtp maxplaybackrate=8000 paramater.

As when FreeSWITCH answers I see;

Media Attribute (a): fmtp:111 useinbandfec=1; maxaveragebitrate=14400; maxplaybackrate=8000; minptime=10

However when I send an Invite offer out I see;

 Media Attribute (a): rtpmap:111 opus/48000/2
 Media Attribute (a): fmtp:111 minptime=10;useinbandfec=1


The OPUS module document seems to suggest we can add this, and if so how can I achieve this, as I have my opus.conf set to;

<param name="use-vbr" value="1"/>
<param name="use-dtx" value="0"/>
<param name="complexity" value="10"/>
<param name="maxaveragebitrate" value="14400"/>
<param name="maxplaybackrate" value="8000"/>
<param name="packet-loss-percent" value="15"/>
<param name="keep-fec-enabled" value="1"/>
<param name="use-jb-lookahead" value="1"/>
<param name="advertise-useinbandfec" value="1"/>


And I also set the SIP profile codec list and absolute codec offer to OPUS at 8000h, and as I say works great when we answer an offer, but not when we originate.

Am I misconfiguring or is this not possible?

I know the RFC states the parameter "maxplaybackrate" is a unidirectional receive-only parameter that reflects limitations of the local receiver, however mod_opus documentation seems to suggest we can set it in the offer, so I should be able to add as;

maxplaybackrate=8000

Is this possible? I just want to ensure I get narrowband end to end.

Many thanks

Jon


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--
Jonathan Hunter
Technical Director /Telephony Developer
M:(+44) 7917 190 438
Email:jhunter at voxboxcoms.co.uk<mailto:Email%3Ajhunter at voxboxcoms.co.uk>

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