I’m having some quality issues - and I don’t really understand why, but what I do undestand is that it’s probably due to my lack of knowledge about SIP and Asterisk. So here are some questions that may be basic to some - but not to me, and for sure also a bunch of other people. Feel free to refer to other sources.
As I understand it, SIP takes care of the dialogue, gives the the pointers where to direct calls, but the actual media transport is left for RTP, and does not pass the SIP server. I have an Asterisk box running as SIP server, but as soon as we get up in call quantity (35+ simultaneos), call quality starts getting bad with delay and jitter. I have another server with the exact same installation and configuration, and taking the same call through this will not result in any quality issue - no other calls are being handled here, and it runs with the same call termination supplier as the other server.
- Does the SIP dialogue influence the quality of the call - if so, how, and why?
- Is it possible to ‘tell’ teh SIP server (Asterisk) not to pass any media through it - if so, how?
- Does the UA have to do with the decision to pass medioa through the server? - in case this actually takes place.
Thanks a lot!