Andrey! Thanks for your interest. I'm Travis, and I work with the engineering teams building Direct Line Speech.
You're correct that the concepts behind Direct Line Speech are a great fit for call center scenarios. There's a huge potential value to application there. At this stage, in preview, we don't currently have built-in support for SIP/RTP to seamlessly integrate telephony I/O, but this is a need that we've heard loud and clear and we're actively investigating solutions as we move forward.
The good news, though, is that you're not blocked on getting started--you can make this work today, just with a little more legwork. Conceptually, you can use an existing telephony endpoint solution (like what Ram's pointing to) as a "middle tier" between your telephony clients and your Direct Line Speech bot. This middle tier service, which could itself actually be a simple bot, would be responsible for handling the RTP/SIP and then forwarding audio between the end user and the "real" bot, using the Speech SDK there to connect with Direct Line Speech. This is a little clunky, to be sure, but so long as the services are co-located within regions, it should still be able to produce a high-quality, low-latency experience the same way you'd get out of a native client.
Thanks again, and please keep sharing your thoughts and questions on the products; we're using your feedback to actively guide where we focus and what we create.