11/11/2023 0 Comments Jitsi sip phoneI propose a in-house Java based Service Creation Environment “SLC SCE”. High Cost of Deployment grade versions.Free versions of the Service Creation Environments do not offer High Availability.Limitations of open source/other market products: Aim :ĭevelop a SCE ( Service Creation Environment ) to addresses all aspects of lifecycle of a Service, right from creation/development, orchestration, execution/delivery, Assurance and Migration/Upgrade of services. Sadly I wasn’t able to complete the job yet I have decided to share a few things about it here. I have traced packets inside the Jitsi-Meet instanceĪnd I can see that packets are sent both ways.I hoped of making a SIP application Development environment a year back and worked towards it earnestly. The logs in Jicofo, Jigasi and Videobridge don’t seem to have any The audio only works from the phone to the browser but not the other With the Jitsi Desktop application I am able to call a phone over PSTNĪnd the audio works both ways. You need to verify in case of one-way audio, where is the point where the audio stops and do not reach the SIP side. Subject: Re: One way audio with Jigasi but works in Jitsi desktop _TRUST_MODE_ENABLED=trueįrom: users [ On Behalf Of Damian Minkov # Activate this property if you are using self-signed certificates or other # The following two props assume we are using jigasi on the same machine as # Should be enabled when using translator mode # Name of default JVB room that will be joined if no special header is included #Sample config with one XMPP and one SIP account configured etc/jitsi/jigasi/sip-communicator.properties: # enable statistics and callstats statistics and the report interval I am including my config the callstats credentials I don't think the problem is with the SIP proxy since audio works when using Jitsi desktop. I have verified that the packets are sent out from the Jitsi host to the SIP proxy, which makes me suspect that there is something wrong with the payload. Users mailing instructions and other list options: I am building the application from master branch sources and my OS Packets sent from the SIP trunk to Jigasi but they are not forwarded to theīrowser. Quite often audio isn’t heard in either end. Is PCMA/8000 and I have tried with both Opus and PCMA/8000 between theīrowser and Jitsi, but it doesn’t make a difference. The codec used between Jigasi and the SIP trunk Have traced packets inside the Jitsi-Meet instance and I can see that Jicofo, Jigasi and Videobridge don’t seem to have any relevant errors. Only works from the phone to the browser but not the other way. However, in my Jitsi-Meet application, the audio With the Jitsi Desktop application I am able to call a phone over PSTN and You need to verify in case of one-wayĪudio, where is the point where the audio stops and do not reach the
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