Bandwidth has multiple geographically redundant signaling proxy facilities in the United States. Please see the specifications below to know what you should expect:
SBC (Session Border Controller) IP Information
Bandwidth employs mated pairs of SBC's for signaling redundancy. For termination services, please ensure that both IP addresses provided are configured for inbound traffic in the event that one SBC is offline. The implementation specialist assisting with your onboarding will provide these IP addresses during the onboarding kick-off call or via the welcome email.
The onboarding specialist will provide a termination rate deck, which will rate each termination call based on the LRN (Local Routing Number) of the dialed number. This provides a LCR (Least Cost Routing) scenario to the customer, which is only possible if the customer is able to perform an LRN 'dip' on each call. If this isn't an option in your internal setup, please contact your onboarding specialist for assistance.
SIP (Session Initiation Protocol)
Bandwidth SIP signaling protocol is designed for RFC3261. If any other method is used, calls will fail to set up.
Allowed Ports for Media/Audio
If a customer PBX is protected by a firewall, you'll need to check the manufacturer warranty to see if the firewall can act as either a SIP ALG or a Back-to-Back User Agent (B2BUA). The following ports are required to allow for full 2-way audio:
- UDP port 5060 – must be opened to support SIP signaling
- UDP ports 1024-64,000 – must be opened either statically or dynamically (ALG) to allow for audio path
- Bandwidth uses multiple IP addresses to allow media from its gateways
The following attributes are allowed within Bandwidth SIP trunks:
- Dial Plans
- Signaling Protocol
- IP Protocol
- Media Anchoring
Bandwidth supports both in-band and out-of-band DTMF outlined in RFC2833.
Bandwidth supports E.164, as well as 10-digit and 11-digit dialing, for outbound calling. E.164 is an internationally recognized standard characterized by a "+" followed by the country code, then phone number. For example:
Local & Long Distance: +19192971000
Bandwidth requires that all SIP and audio be delivered via UDP, in packets no larger than 1350 bytes. TCP is not supported.
The following codecs are supported by Bandwidth:
- G711ulaw, G729a, ILBC (will default to ptime 30)
- G711ulaw, G729a (will default to ptime 20)
Call Concurrency Limits
Please contact the Bandwidth support team to determine what the appropriate call concurrency limits are. This will be calculated by type of traffic and expected MOU. If a customer's call concurrency limit is reached, a 503 Service Unavailable or 486 Busy Here signal message will be provided to the customer.
Redundant Capacity: Bandwidth will allocate capacity on the following Outbound Calling rate decks to allow them to be capacity redundant during scheduled maintenance events: Flat Rate, Priority, Hybrid, On-Net, and Local.
Non-Redundant Capacity: Bandwidth will allocate non-redundant capacity on the following Outbound Calling rate decks: Wholesale, Carrier, and Select. Please note: in the event of a schedule maintenance or a service impacting event, you should anticipate and expect decreased capacity and route advance to other vendors.
Supported Caller ID / Privacy Types
- FROM field – default option for caller ID name, number and rating
- RPID (Remote Party ID) – secondary option
- P-‐Asserted ID – supported option
- Privacy Headers – supported option
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