start_dtmf_generate is not working with opus code
Hi all, I am facing a issue with start_dtmf_generate when i use opus codec and SIP INFO dtmf. Caller ( opus/dtmf ) ------------------------ Freeswitch ----------------------------- Calleee ( ulaw/inband) Caller is using jssip webrtc and dtmf is in INFO. For inband dtmf generation i am using start_dtmf_generate in caller channel. The setup is working when i use ulaw or ilbc or alaw codec. When i use webrtc with OPUS codec i am seeing the inband dtmf generation is not working. Webrtc with ulaw codec and dtmf INFO is working fine. Version : FreeSWITCH Version 1.6.17~64bit ( 64bit) I am attaching the opus.conf.xml for the reference. Any idea, why start_dtmf_generate is not working with opus codec ?. Do i need to set any setting for opus ?. Regards
inband tones are not reliable when using complex codecs. You also generally do not want to do inband at the same time as another dtmf method like info.
On Jan 29, 2018, at 3:23 AM, Andrew Paul <andrew.paul85@gmail.com> wrote:
Hi all,
I am facing a issue with start_dtmf_generate when i use opus codec and SIP INFO dtmf.
Caller ( opus/dtmf ) ------------------------ Freeswitch ----------------------------- Calleee ( ulaw/inband)
Caller is using jssip webrtc and dtmf is in INFO. For inband dtmf generation i am using start_dtmf_generate in caller channel.
The setup is working when i use ulaw or ilbc or alaw codec. When i use webrtc with OPUS codec i am seeing the inband dtmf generation is not working. Webrtc with ulaw codec and dtmf INFO is working fine.
Version : FreeSWITCH Version 1.6.17~64bit ( 64bit)
I am attaching the opus.conf.xml for the reference.
Any idea, why start_dtmf_generate is not working with opus codec ?. Do i need to set any setting for opus ?.
participants (2)
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Andrew Paul
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Michael Jerris