I have a high latency problem between a softtone I made and asterisk. it always hovers over 1000 milliseconds (according to those who have installed the switchboard and too high) … the Sotftone is made in java and uses a library that I downloaded on the internet. Doing the tests with the example provided by the library itself, I noticed that, unfortunately also in this case, the latency is high.
I have two questions to ask:
- If the latency is high what consequences do I have in terms of call quality?
- what can I check, on asterisk and on my code, so that I can make some optimization / improvement?
I hope I don’t have to throw away all the work done
below is an example from asterisk with sip show peers
dev222/dev222 OK (1026 ms) my softone
dev224/dev224 OK (111 ms) 3cx phone
Assuming this applies to media, as well as signalling, of course it does. When there was significant use of geostationary satellites only one hop was normally allowed, even though the the total delay was only 540ms, two ways with one hop. Even one hop did, I believe, make conversations difficult.
This is a network (or possibly an overloaded host) problem. To a first approximation there is nothing you can do in the clients. However, if you own the network, you may be able to configure it prioritise certain traffic, and you can configure Asterisk to mark some traffic as high priority. You will need to do the same on the phone.
If this is on a mobile phone, you may be seeing a signalling only problem due to the app being put into a power saving mode.
you should set keepalive on both asterisk an your softphone
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