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Asterisk SIP Call audio delay - Asterisk Support ...
https://community.asterisk.org/t/asterisk-sip-call-audio-delay/30669
Asterisk SIP Call audio delay. Asterisk. Asterisk Support. system February 26, 2010, 9:24pm #1. System info: Asterisk 1.4.28 Phones Polycom IP650 and Spectralink 8002. I’ve seen this posted in numerous other areas but no fix shown. A good example of what we’re experiencing is described here:
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
The Asterisk server is behind double NAT in a local LAN. The phones are in the same LAN. I’ve multiple sip accounts and sip providers. Problem: If I receive a call on one of my sip accounts, the audio from the external caller to the local phone is delayed by about 2s. Audio from local phone to external caller is not delayed!
How to measure audio delay between two Asterisk …
https://stackoverflow.com/questions/22302113/how-to-measure-audio-delay-between-two-asterisk-channels
audio delay asterisk. Share. Improve this question. Follow asked Mar 10 '14 at 13:53. user3401988 user3401988. 1. Add a comment | 1 Answer Active Oldest Votes. 0 You can use Echo application on one end and send Beep on other end. After that you can measure delay between beeps. ...
Asterisk Call Delay - Asterisk SIP - Asterisk Community
https://community.asterisk.org/t/asterisk-call-delay/78920
I have a strange issue when some calls have 10-15 seconds until Asterisk is sending INVITE to the peer. The problem is not permanently, few calls are working fine, and then 1-2 calls are with delay. For ex. one call with delay: CLI> pjsip show history. 00003 1552335791 * <== 192.168.2.200:56089 INVITE sip:[email protected] SIP/2.0
Audio Buffer between core and channel driver - Asterisk ...
https://community.asterisk.org/t/audio-buffer-between-core-and-channel-driver/90327
I have been looking through the functions and applications, and have not seen exactly what I am searching for, so I thought I would ask here. Is there any implementation of an audio buffer / Delay in asterisk? This buffer would take voice frames buffer them up for a set amount of time, and then release them to the next destination. If not, is there a application or …
audio - Is a short delay before I can hear the person I'm ...
https://serverfault.com/questions/109872/is-a-short-delay-before-i-can-hear-the-person-im-calling-a-known-issue-with-a
Whenever I call someone using my workplace's Asterisk VOIP server there's a 2-3 second delay before I can hear them so I always miss them saying "hello". This happens with any SIP client running on windows, Mac, or linux. After that the call goes fine. It only happens for outgoing calls - the audio for incoming calls is fine.
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