Dears,
When l place a call inside a Meet Conference the call is completed but no audio sound is heard on Meet Conference. I am able to hear audio sound on the mobile side.
I have grabbed a tcpdump capture from an audio voip on Jitsi Server. It is possible to hear both sip audio directions on Wireshark. What could be happening? Why the sound coming from PBX towards to Meet Conference can not be heard?
By the way, both sides negotiate PCMU/ULAW codec.
Do you see media coming from sip side to jigasi? Do you see media from jigasi to the bridge on port 10000?
I placed a call from Meet Conference to a Mobile Phone.
I grabbed a packet capture form pcap jigasi logs directory.
I am able to see following RTP Streams on Wireshark.
PBX Port 18270 >>> OPUS Codec >> Jigasi Port 10002
Jigasi Port 10002 >> OPUS Codec >>> PBX Port 18270
Jigasi Port 10001> RTPType-111 >> VB Port 10000
As told you before, i can hear sound on Mobile Phone but cant hear Sound on Meet Conference.
Are you testing with chrome? We got some reports about problems with Safari and Firefox which we haven’t still look at.
I’m testing with Chrome.
By the way, i have a found a specific scenario where i can hear the voice on Meet Conference.
When i include another smartphone app jitsi-meet member to the conference, i can place a call to a mobile and both audio directions is played. Weird.
Which jigasi and jitsi-meet do you use?
Can you send jigasi ligs and web js console logs from non working case?
dpkg -l|grep “jigasi”|tee -a
ii jigasi 1.1-166-g929a439-1 amd64 Jitsi Gateway for SIP
dpkg -l|grep “jitsi”|tee -a
ii jitsi-meet 2.0.5390-3 all WebRTC JavaScript video conferences
ii jitsi-meet-prosody 1.0.4628-1 all Prosody configuration for Jitsi Meet
ii jitsi-meet-turnserver 1.0.4628-1 all Configures coturn to be used with Jitsi Meet
ii jitsi-meet-web 1.0.4628-1 all WebRTC JavaScript video conferences
ii jitsi-meet-web-config 1.0.4628-1 all Configuration for web serving of Jitsi Meet
ii jitsi-upload-integrations 0.15.15-1 all jitsi-upload-integrations
ii jitsi-videobridge2 2.1-416-g2f43d1b4-1 all WebRTC compatible
Selective Forwarding Unit (SFU)
For a non-working case:
Jigasi Logs: 2021-02-17 18:15:10.014 INFO: [74] org.jitsi.jigasi.xmpp.CallControl.handleDialI - Pastebin.com
Web JS Console Logs: Content.js:122 @atlaskit/modal-dialog: Deprecation warning - Use of the footer p - Pastebin.com
Yeah I took a look but nothing obvious and for the moment I’m not sure how to proceed. Maybe doing a tcpdump on jigasi and see is media received from the sip side and is it forwarded to the bridge.
I did a tcpdump just like this: tcpdump -i any udp port 5060 or udp portrange 10000-65000 -s 0 -w jigasi_tcpdump.pcap. Take a look at the Lost packets from Jigasi to PBX.
@damencho I’ve figure it out that the issue is regarding Chrome browser itself. If i place a sip call inside a Conference using a Jitsi App on my smartphone, both sides can hear themselves. Weird, isn’t it? So, how can i solve itt?
Regards
How do you place a sip call using the jitsi app on your phone?
@damencho R: The same way i place on chrome browser, just inviting a new person by its mobile phone number.
Yep that is strange …
From the browser you don’t hear jigasi, what happens if you open 3 tabs? Do you hear the audio coming from other tabs?
@damencho Sorry for the delay.
On the third Tab i can not see audio “leds”. On the other tabs i am able to see it.
@damencho Is it a good a idea opening an issue request on Jitsi Issues · jitsi/jigasi · GitHub ? What do you think about? ANy ideas?
Regards
These are 3 tabs in the browser and one of them does not have audio coming? This is strange … and there is no jigasi in this situation, right?
