2014 2017
VoIP Infrastructure & Carrier Interconnect
The shift from circuit-switched PSTN to packet-switched
VoIP (Voice over IP) was the defining technology transition
of this era, and Joomo was in it from day one. We built out a carrier-grade
VoIP core using Asterisk PBX and FreeSWITCH
as session border functions, handling call signalling via
SIP (RFC 3261) with full
SDP (Session Description Protocol) offer/answer negotiation
for codec selection supporting G.711 (PCMU/PCMA), G.729, G.722 wideband,
and Opus for high-definition voice.
Media transport ran over RTP (RFC 3550) with
SRTP (RFC 3711) for encrypted media streams, using
DTLS-SRTP (RFC 5763) for key exchange the same
mechanism now powering WebRTC endpoints in CallJots. NAT traversal was
solved through a full STUN / TURN / ICE stack, allowing
clients behind enterprise firewalls and carrier-grade NATs (CGN) to
establish direct media paths. Call quality was monitored using
RTCP metrics jitter, packet loss, and
MOS (Mean Opinion Score) scoring feeding QoS dashboards
used by interconnected carriers.
Carrier interconnects were provisioned over
SIP trunks with Tier-1 operators, with call routing
governed by LCR (Least Cost Routing) logic that factored
in per-destination termination rates, codec compatibility, and real-time
quality scores from RTCP feedback.
SIP/SDP
RTP/SRTP
DTLS-SRTP
STUN/TURN/ICE
Asterisk
FreeSWITCH
MOS/QoS
LCR