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No audio on answer #73
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Also, it doesn't work with the pre-built binaries from here. Tried with universal 1.3.0 |
@oksakhartman
Not sure if related, but pjproject-git 2.13.r79.gf60d1c4-1 - built with #define PJMEDIA_CONF_USE_SWITCH_BOARD 1 I can paste the whole Here's the configure command
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@oksakhartman I've managed to test some scenarios :) A - tg2sip (and Asterisk) H -> A -> B - no audio Not sure, why few of the calls had audio (also, 7-8 in a row), between H -> A -> B ? I'm starting to suspect, that if the call gets "routed" through "udp reflector" that "accepts" version 2.4.4 - it works :/ Here are the changes that I've made to get branch feature/libtgvoip-update building without errors using tdlib commit id
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@oksakhartman Yup, few calls had audio. Maybe 10 out of ~200 :) Also, one time it worked for 5-6 calls made in a row, then after restarting tg2sip doesn't worked anymore (still not worked from that time) :/ Looks like a bug, yeah ... If it was a "rejection" from deprecation, it shouldn't worked at all :) |
@oksakhartman
Can you elaborate more on that? Where, how, any hints/tips/steps would be helpful! :) |
I will be grateful if you provide compiled x64 binaries for tests |
We tested - no voice :( |
No voice :( |
Что значит вроде, вы тестировали? |
Протестировали, голоса нету. |
Hi there, i'm having troubles getting the audio rtp streams working, it doesn't "starts" 99% of the times.
The SIP signalling works, I'm trying a SIP->TG call, the mobile phone (tg) rings, but on answer - there's no audio.
Asterisk - on real IP (in the logs - 192.168.1.6 is the internal address in the DMZ)
tg2sip (running branch feature/libtgvoip-update) - using opus (running on the same machine)
I've saw few times
First audio packet - setting state to ESTABLISHED
and then the audio started. As far as I understand, there's no STREAM_DATA* packet for some reason?!Not sure if it's a networking/latency issue on my side or something else? Any hints are welcome :)
I've tried with
jitter_initial_delay_60=4
- it worked 10 times in a row, after restarting tg2sip doesn't worked anymoreptime 20, instead of 10 (in sip.cpp)
ep_cfg.medConfig.audioFramePtime = 20;
ep_cfg.medConfig.ptime = 20;
No audio
With audio
No audio log
Audio wokring
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