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Adaptive Delay Buffer (2.10) - PJSIP
https://www.pjsip.org/pjmedia/docs/html/group__PJMED__DELAYBUF.htm
Create the delay buffer. Once the delay buffer is created, it will enter learning state unless the delay argument is specified, which in this case it will directly enter the running state. Parameters Returns PJ_SUCCESS if the delay buffer has been created successfully, otherwise the appropriate error will be returned. pjmedia_delay_buf_put ()
Audio Latency | pjsip blog
https://blog.pjsip.org/tag/audio-latency/
PJSIP 0.9 is Released: Audio Latency, TURN implementation, IPv6, G.722, and More. Finally, after months of delay, PJSIP version 0.9.0 is released. This has been the longest gap (8 months) between releases, and consequently it has the most modifications in it (there have been 100+ tickets done on this release).
Audio delay of 2s for incoming calls - "Answer() before ...
https://community.asterisk.org/t/audio-delay-of-2s-for-incoming-calls-answer-before-dial-fixed-it/85711
; extension with 2s delay for incoming pstn-call [sipprovider-in] exten = _X.,1,NoOp(Provider 1) same = n,Dial(PJSIP/201,20) same = n,Hangup() Here is the log for the delayed example: – Executing [xxxxxxxxxx@sipprovider-in:2] Dial(“PJSIP/sipprovider-0000003a”, “PJSIP/201,20”) in new stack – Called PJSIP/201 – PJSIP/201-0000003b is ringing
Workaround for frame bursts and drifts from audio ... - PJSIP
https://trac.pjsip.org/repos/ticket/438
it adds audio latency of up to PJMEDIA_SOUND_BUFFER_COUNT frames, while theoretically the delay should be limited to the maximum burst level of the device. it will add even more end to end latency when the sound device is connected directly to the stream (rather than via conference bridge), because of how the jitter buffer is implemented.
[pjsip] Audio problem
https://pjsip.pjsip.narkive.com/PZQgcBbG/audio-problem
sound-problem page of pjsip, I do hear the sound. So it has nothing to do with the pjsip libraries, but something else. So my device is definitly working properly, the pjsip executable is working, so that are all not the problem. So next I will copy the program output and the way I initialize pjsua and play the audiofile.
Asterisk 13 Configuration_res_pjsip - Asterisk Project ...
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_res_pjsip
This may result in a delay before an attack is recognized. ... chan_pjsip will send a 183 Session Progress when told to indicate ringing and will immediately start sending ringing as audio. If set to no, chan_pjsip will send a 180 Ringing when told to indicate ringing and will NOT send it as audio.
pjsua_media_config Struct Reference (2.10)
https://www.pjsip.org/pjsip/docs/html/structpjsua__media__config.htm
unsigned pjsua_media_config::max_media_ports. Specify maximum number of media ports to be created in the conference bridge. Since all media terminate in the bridge (calls, file player, file recorder, etc), the value must be large enough to support all of them. However, the larger the value, the more computations are performed.
HD video is very slow. · Issue #112 · VoiSmart/pjsip ...
https://github.com/VoiSmart/pjsip-android/issues/112
davallclarke commented on Jul 27, 2020. Any video resolution 1280 × 720 or higher is very slow. The audio plays 5-10 seconds before the video. Not only does it have high latency but many frames are missing or the video plays in slow motion. I modified the config below in 100s of combinations but no significant improvement occurred.
sip - How to get the audio stream from PJSIP when there …
https://stackoverflow.com/questions/46243029/how-to-get-the-audio-stream-from-pjsip-when-there-is-no-audio-hardware-device
I want to use PJSIP's C API to record the incoming audio to a file on a machine with no hardware sound device . I'm unsure about the details, but the sparse documentation for PJSIP suggests it should be possible through the pjsua_set_null_snd_dev() call.
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