Role of SIP trunk provider

I want to understand what exactly happens on SIP trunk provider side.

I’m going to host Asterisk system here inside our office. All phones will be on a LAN. Asterisk system will be virtualized running on ESXi and will be connected to SIP provider outside. For our load and purpose I don’t see how internal setup can be any better.

I want best possible voice quality. I was wondering what actually happens so I can choose SIP provider wisely. Do they actually proxy data packets through their servers? Do they have to support specific codecs? Do I need to make sure their servers close by?

I’m not sure if this forum ok to ask for suggestions but I will need to pick one. Right now we have 3-4 employees using system and rarely need more than 2 channels at the same time. We don’t do much calling either. But our system need to sound clear and when we do use it - it should be good. What it means at this point - I’m ok with paying slightly higher rate if needed since it’s not going to cost much anyway…

I have RingCentral account right now and this is NOT what I want quality-wise.

Our internet is business fiber with 25/5 and those speeds steady. Sorry I don’t know about jitter…

If speech quality is so critical, why are you running Asterisk on a virtual machine?

I assume you are talking about ITSPs (Internet Telephony Service Providers) as SIP doesn’t really have a concept of a trunk, and nothing more than IP connectivity is needed for trunked SIP to SIP communication.

Basically they will have the equivalent of a PABX with both ISDN and SIP connectivity. Some small ones may simply use Asterisk, but larger ones are likely to use things like SIP proxies to do load balancing.

In SIP both peers always constrain the choice of codecs Hopefully they will offer the codec used by the PSTN in your part of the world, as one choice (G.711 mu-law or A-law).

Going back to speech quality, if that is critical, you want to have the ISDN interface on your site, and not rely on the public internet or an ITSP.

I’m sorry, just learning all this stuff, some things I’m saying may sound strange, thats because I try to understand…

  1. Why running asterisk in VM is a problem? Hardware will be good, SSD drives, etc. It is not running on desktop in VirtualBox, I plan to run it on hypervisor (VMWare ESXi)

  2. We have couple static IP’s here. Will it be beneficial if Asterisk will be connected to internet on it’s own static IP port? Box won’t be on our company network than…

  3. So, ITSP may be slow at connecting peers, may fail doing so, etc. But it will NOT affect quality of a call, do I understand correctly? It will help both ends to establish connection but than it will depend on those ends and ITSP is out of the game during call, correct? We can try for best codec from our side but it will be up to other side to decide, right? What if we call to landline?

  4. Yes, we want speech quality but having ISDN is out of question. We are small company. I just want to get best out of our setup…

By ISDN, do you mean we need to get phone provider to bring line in and we can have special card in Asterix box to drive it? Is that what you mean? So our calls go on a real phone line? And we would have all the features of Asterisk?

VM’s have poor timing integrity and are likely to increase jitter.

The media has to pass through the ITSP as it has to convert from packet switched (SIP) to circuit switched (ISDN).

Ok. If I just get plain cheapest phone line from a phone company and buy a card for asterisk.

Will this give me 1 channel with a clear voice? I might loose caller id (if don’t subscribe), but overall I will have fast and good line. Do I understand correctly?

And if I expose * to internet - I will be able to use soft phones from anywhere to dial out using office system.

If my assumptions correct - what card is that? (name, terminology)?