At HP, we are heavy users of conference calls. Most of those calls are provided by InterCall, and we are moving over to using Lync running on the corporate network. And when I was at RedHat, we used a lot of InterCall there as well. And of course, there are all the various external partner calls on Cisco WebEx and on Join.ME.
I have developed a wishlist of features for conference call service providers.
My first simplest wish is that bridge id's / meeting id's should be forbidden from having a repeating digit. This is big pain point that expresses itself when dialing quickly or when using an automated dialing string. I would say that at least half the time, when there is a repeated digit dialed quickly, it gets mis-interpreted.
Next, I wish there was a simple modern web app (with an underlying REST API) that lists all the lines connected into the bridge, showing for each line the caller id, registered user, connection time, is-muted, and instantaneous momentary current sound input level. By "registered user", I would like people to be able to say "when you see this CID/ANI, that's ones of my phones, and display my name". And for the sound level, it would be "who's line has the barking dog"?
And my big "I wish for a pony" wish: I wish for standards compliant SIP VoIP interfaces. Often I connect to a concall by running a SIP client on a machine or device, connecting that SIP client to a SIP/PSTN gateway service provider, and then dialing into one of the PSTN numbers of the concall service provider. It seems that it would be better to bypass some of that complexity, and just be able to point my SIP client directly to sip:firstname.lastname@example.org
Intercall actually does provide a HA QoS MPLS SIP interface designed to be plugged into large corporate soft PBX systems, but it's not available over the public internet. It is understandable that Intercall can't guarantee call quality for internet VoIP, but a supported best-effort interface would still be useful.
Microsoft Lync is, of course, VoIP, but it's not at all interoperable with any other VoIP client. It has "embraced and extended" SIP with a lot of semi-documented MSFT-only extensions. It does use well known codecs, but it runs media transport over TCP in a way that is in no standard and like nobody else does. (The term "realtime media transport over TCP" makes me facepalm).