Outbound calling integration guideFollow
Bandwidth has multiple geographically redundant signaling proxy facilities in the United States. Please see the specifications below to know what you should expect.
SBC IP information
Bandwidth employs mated pairs of Session Border Controllers (SBCs) for signaling redundancy. For termination services, please ensure that both IP addresses provided are configured for outbound traffic in the event that one SBC is offline.
Note: 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 an 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.
Bandwidth Session Initiation Protocol (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 your 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 the 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 isn't currently 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 your Bandwidth Support Team to determine what the appropriate call concurrency limits are for your account. This will be calculated by the type of traffic and expected MOU (Minutes Of Use). If your call concurrency limit is reached, you'll be provided a 503 Service Unavailable or 486 Busy Here signal message.
Bandwidth will allocate capacity on the following Outbound Calling rate decks to allow them to be capacity redundant during scheduled maintenance events:
- Flat Rate
Bandwidth will allocate non-redundant capacity on the following Outbound Calling rate decks:
Note: In the event of a scheduled 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 is the default option for caller ID name, number, and rating
- RPID (Remote Party ID) is a secondary option
- P-Asserted ID is a supported option
- Privacy Headers is a supported option
Was this article helpful?
4 out of 5 found this helpful