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

Oancea, Dragos dragos.oancea at vonage.com
Wed Jul 12 15:14:36 UTC 2017


Hi Jon ,

So the fmtp that goes to the carrier is the same as the one sent by Janus ?
Try : <param name="inbound-codec-negotiation" value="greedy"/> on the SIP
profile. I assume you have this set to "generous".
Also please use FS master or the latest release.
If it still does not work please open a FS jira.
You see the fmtp being changed with the XML values when you originate the
call through fs_cli, right ?

Cheers,
Dragos


On Wed, Jul 12, 2017 at 3:33 PM, Jonathan Hunter <hunterj91 at hotmail.com>
wrote:

> 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>
> 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>
>
>
> 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>
> 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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>
>
> _________________________________________________________________________
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> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
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>
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>
>
>
> --
> Jonathan Hunter
> Technical Director /Telephony Developer
> M:(+44) 7917 190 438 <+44%207917%20190438>
> Email:jhunter at voxboxcoms.co.uk
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
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>
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