For international video conferencing, I’d evaluate Tier-1 providers by actual paths from your offices to the conferencing provider’s media edges, not by a generic “best backbone” ranking. Packet loss and jitter can be caused by the access ISP, peering, transoceanic path, congestion, or your own VPN/firewall—not necessarily the Tier-1 core.
Tier-1 providers worth shortlisting
| Provider | Network characteristics relevant to real-time video | Where I’d investigate it |
|---|
| Arelion (AS1299) | Very extensive direct interconnection; its backbone spans North America, Europe and Asia, with 350+ PoPs. Arelion says its network reaches 95% of North American/European end users within one hop. Arelion Arelion | Particularly interesting for US↔Europe and Europe↔Asia traffic where reducing intermediary hops matters. |
| NTT Global IP Network (AS2914) | Global Tier-1 backbone across the Americas, Europe, Asia and Oceania. NTT publishes a backbone SLA of ≤0.1% average monthly packet loss and ≤250 µs average jitter. NTT Gin NTT Gin | A strong candidate for Asia-Pacific/trans-Pacific paths and globally distributed enterprises. |
| Tata Communications (AS6453) | Tier-1 backbone with extensive subsea infrastructure, 250+ PoPs according to its 2026 materials, and 300+ global peerings. Its own case study reports a ~15% latency improvement on an Asia–US route after optimizing its backbone. Tata Communications Tata Communications | Worth testing heavily for India/APAC↔US/Europe, Africa and other long-haul international paths. |
I would also put Lumen, Cogent and other major global transit providers into an RFP/measurement exercise, but I wouldn't declare one universally superior without seeing your source/destination pairs. The important variable is which provider has the cleanest path to the particular Teams/Zoom media infrastructure you're reaching.
The bigger issue: don't optimize only for “Tier-1”
For Teams, Microsoft explicitly recommends keeping the path from the endpoint to Microsoft 365 short and direct, using local Internet egress close to the user. It also recommends avoiding VPN hairpinning for real-time media because it can introduce additional latency and jitter.
Microsoft's network-quality measurements specifically consider UDP latency, UDP jitter and UDP packet loss. Microsoft Learn As a practical target, Microsoft documents <300 ms RTT, <30 ms jitter and <1% packet loss for optimal real-time media conditions in its Teams guidance.
So I'd test the providers this way:
- Identify the actual conferencing media destinations for each region rather than testing arbitrary Internet endpoints.
- Measure UDP loss, jitter, RTT and burst loss, separately for US↔Europe, US↔Asia, Europe↔Asia, etc.
- Run the same tests through each prospective carrier's PoPs.
- Examine BGP paths and submarine/long-haul segments to see where paths diverge.
- Test during peak business hours for at least several days; a 30-minute speed test can completely miss intermittent congestion.
- Prefer diverse dual-carrier connectivity rather than putting all real-time traffic on one Tier-1.
- Apply QoS end-to-end. Microsoft notes that QoS needs to be consistently applied across the managed path; otherwise an unconstrained segment can still introduce loss/jitter. Microsoft Learn Zoom likewise supports DSCP classification, with separate default markings for audio, video and screen sharing.
If your primary problem is packet loss rather than latency, I would especially investigate NTT and Arelion first, alongside Tata for Asia-heavy traffic—but treat that as a shortlist for measurement, not a universal ranking. NTT is particularly notable because it publishes explicit packet-loss and jitter SLA figures, which gives your engineering team something concrete to put into an SLA comparison.
If you give me your office regions/cities (e.g. New York → London → Singapore → Tokyo), conferencing platform (Teams/Zoom/Webex), and current ISP, I can turn this into a carrier-by-carrier test matrix with the BGP/latency/loss metrics I'd collect.
For Teams, Microsoft explicitly recommends keeping the path from the endpoint to Microsoft 365 short and direct, using local Internet egress close to the user. It also recommends avoiding VPN hairpinning for real-time media because it can introduce additional latency and jitter.
Microsoft's network-quality measurements specifically consider UDP latency, UDP jitter and UDP packet loss. As a practical target, Microsoft documents <300 ms RTT, <30 ms jitter and <1% packet loss for optimal real-time media conditions in its Teams guidance.
So I'd test the providers this way: