Inbound calling integration guide
FollowBandwidth 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 inbound calling services, please ensure that both IP addresses are configured for inbound traffic in the event that one SBC is offline. Your implementation specialist will provide these during the onboarding kick-off call or via the welcome email.
SIP
Bandwidth's Session Initiation Protocol (SIP) is designed for RFC3261. If any other method is used, calls will fail to set up.
Allowed ports for media/audio
If your PBX is protected by a firewall, you'll need to verify the manufacturer's compliance to ensure 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 1024-65534 – must be opened either statically or dynamically (ALG) to allow for the audio path.
- Bandwidth uses multiple IP addresses to allow media from its gateways.
Attributes
The following attributes are allowed within Bandwidth SIP trunks:
- DTMF
- Dial Plans
- Codecs
- Signaling Protocol
- IP Protocol
- Media Anchoring
Supported DTMF
Bandwidth supports both in-band and out-of-band Dial-Tone Multi-Frequency (DTMF) outlined in RFC2833.
Dial plans
Bandwidth supports E.164, 10-digit, and 11-digit dial plans for inbound calling.
Note: Some inbound products only support E.164. If your equipment can only support 10-digit or 11-digit dial plans, please let your implementation specialist know. This will help ensure that you select a product that meets your needs.
E.164 is an internationally recognized standard characterized by a "+" followed by the country code and the phone number. For example:
Local and Long Distance: +19192971000
International: +4402074942020
IP protocols
Bandwidth requires that all SIP and audio be delivered via the User Datagram Protocol (UDP), in packets no larger than 1350 bytes. Transmission Control Protocol (TCP) isn't currently supported.
Supported codecs
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 open a ticket with your Bandwidth Support Team to determine what the appropriate call concurrency limits are. If you reach your call concurrency limit, you'll receive a 503 Service Unavailable or 486 Busy Here signal message.
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
Questions? Please open a ticket with your Bandwidth Support Team or hit us up at (855) 864-7776!
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