How to test VoIP throughout Russia?

Question

Asterisk

While developing a call automation service, we came to the conclusion that the quality promises of an IP telephony provider are just words. Packet delay, DTMF recognition, Caller ID pass should be tested. The development team tried all the local telephony operators and telephone sets, but from time to time there are complaints from customers: either the Caller ID is not the same, or DTMF is not recognized.
1. Should I hope for stability when working through SIP?
2. Is it possible to gather a community of testers throughout Russia (maybe someone has already collected)?
3. What factors, independent of the SIP provider, really affect the stability of the work?

Answer 1

Often, problems with the transmission of caller-id, cdpn, cgpn, sdp appear at the points of contact between equipment from different vendors, because there are recommendations rfc 3261, 2833 and often not all manufacturers strictly follow these recommendations (or misinterpret them). Therefore, SIP is ideally good, but in practice it all depends on the manufacturer.
Maybe I'm wrong, of course. It would be better if someone corrects me.

Answer 2

The quality of SIP does not depend on your SIP operator, but on the quality of the Internet channels they use. In the Russian Federation - there are several high-quality providers

Answer 3

Still, as it turned out in practice, it depends on the settings of local telephony providers.
A specific example with CallerID:
We set our real number as CallerID on the SIP account. And after that, calls from this sip-account to all numbers of the region (area) stopped going through. As it turned out later, Rostelecom's "ATS" is configured in such a way that it does not receive external calls from numbers belonging to its internal capacity, which is generally logical. I had to use a sip-number without CallerID to make calls to the cities of our region.
And yes, as we wrote earlier, most of the problems arise when ip-traffic lands on telephone networks.