I’ve been running Asterisk pretty successfully for a few years now. Successfully in the sense that I have it configured to talk to my Linksys SPA3102 for making outgoing calls and receving incoming calls from my local telco operator. I also have a DID number in the Netherlands which I still use for my visits there on business and also to talk to friends. Other international calls I route out through various VoIPs. And all is routed via Asterisk using a Linksys SPA941 “hard ip phone” and a pap2t connected to a cordless DECT phone. So all was fine?
Well no. I’ve been plagued for months with an issue I couldn’t resolve. My wife complained (rightly) that the phone calls had an unnaturally long delay when talking to people which made the conversation much more difficult. If you tried to interrupt someone who was talkng they didn’t hear you immediately and this broke up the flow of conversation completely.
So I looked at the Asterisk config and tweaked this and that. It didn’t seem to help. I looked on Google for complaints from others of similar symptoms but to be honest couldn’t really find the cause. There were some comments that referred to Asterisk RTP config being inflexible and requiring making adjustments on the ATA I was using but to be honest this didn’t make much difference.
I’ve been tempted to try and switch to another VoIP product like Freeswitch but that’s harder to configure (mainly through lack of experience with it, even if it looks promising).
Finally it seems (and I’m still not 100% sure), though the signs are promising, that the problem is something as simple as the SPA3102, PAP2T and SPA941′s default configuration having a high network jitter configuration. This is to improve things on an internet based voip call. However this adds loads of extra latency and if the phones are connected directly to Asterisk this default configuration adds a lot more latency than you would expect: latency from the phone to asterisk and then latency from the phone to the PSTN number (via the SPA3102), the combination being enough to destroy the usability of the call.
I’ve changed this now to low network latency and the results are promising, but I’m surprised not to have seen this problem mentioned in the Asterisks books I’ve bought, nor various forums and mailing lists I’ve been following.
So if you find that you have real problems with latency and you think it may be the pc, or it may be Asterisk then take a look at the soft or hard phones you are using and check the jitter configuration, at least on the Linksys products. This might make a huge difference and may your system usable.