Voice Insights is a call data analytics engine that provides details ranging from real-time metrics for individual calls to account-wide historical trends and aggregates to assist in diagnosing call performance problems, investigating call quality reports, and identifying emerging issues. Voice insights already includes WebRTC and Carrier Insights and today, we are excited to announce that Elastic SIP Trunking and SIP Interface Insights are now also available, in Beta. You can now easily have a deep understanding of your calls when they traverse your SIP communications infrastructure. These insights enable you to have an end-to-end understanding of call quality, no matter how the call is made or received, and help diagnose call establishment and performance issues.
Using Twilio SIP Insights data, for every SIP call made or received through Twilio, you can analyze call performance from an aggregated view down to granular details about a call. For any given call you can easily see relevant network metrics (Jitter, Packet Loss), SIP events and other related information (Post Dial Delay, Who Hung Up, Region, IP addresses, etc.) allowing you to pinpoint the root cause of issues.
Know your SIP calls
Monitor your SIP calls and rapidly see key information:
- Elastic SIP Trunking
Properties: Last SIP response code, post-dial-delay, who-hung-up, call state & Trunk SID.
SIP-leg: User agent, SIP-Call-Id, IP addresses for signalling and media, incoming Region, last SIP response code, post-dial-delay, who-hung-up, call state.
Carrier-leg: Carrier, device and country. - SIP Interface
Properties: Last SIP response code, post-dial-delay, who-hung-up & call state.
SIP-leg: User agent, SIP-Call-Id, IP addresses for signaling and media, incoming Region.
SIP insights now makes it easy for you to troubleshoot issues like one-way audio issues, successful vs. failed call completions, by providing you data on signaling and media IP addresses that are actually used. Visibility into these metrics allows you to troubleshoot your calls and diagnose emerging issues faster.
Visualize SIP call events and metrics
For every SIP call you will have visibility into the metrics on the RTP stream sent to and received from your SIP infrastructure: jitter, packet loss, and packets sent/received. Elastic SIP trunking calls will include metrics for both the SIP and Carrier legs. With this information, you can better assess the quality of a conversation and determine the side of the call that might be performing poorly.
Getting Started
SIP Insights can be enabled by checking the Carrier/SIP Insights box in Twilio Console. If you’ve already enabled Carrier Insights, SIP Insights are included.
Pay-as-you-go Pricing
On-demand pricing for Carrier and SIP Insights available:
Carrier calls: $0.0025 per call minute
SIP Interface calls: $0.0025 per call minute
Elastic SIP Trunking calls: $0.0040 per call minute (includes both SIP & Carrier legs)
and you have the option to do an annual commit to take advantage of volume pricing.
Learn more about pricing here.
Questions? Please reach out to abpelaez@twilio.com
We can’t wait to see what you build!
Announcing Voice Insights for Elastic SIP Trunking and SIP Interface