UC trunking integration guideFollow
Bandwidth has multiple geographically redundant signaling proxy facilities in the United States. In this section we'll guide you to how to set up your service and what you should expect.
Bandwidth employs a mated pair of IP proxies for signaling redundancy, so please ensure that both of them can be accessed for that purpose.
To obtain the specific IPs, please reach out to your Implementation Specialist, if you're currently working with one. If not, please open a ticket with your Bandwidth Support Team.
- For Origination (inbound), please ensure that both IP addresses are whitelisted as signaling could originate from either proxy.
- For Termination (outbound), please ensure that both IP addresses are configured for outbound traffic in the event that one is offline.
Bandwidth doesn't support REGISTRATION. The customer is required to provide a Public Static IP Address for Bandwidth to send calls to. For the customer to send Bandwidth calls, the user needs to provide the Public Static Address. If a Registration method is sent, Bandwidth may return a 200 OK message, but this isn't guaranteed.
Bandwidth will only allow signaling via the SIP method encompassed in RFC 3261. If any other protocol is sent, calls won't set up.
Ports to be opened
Many customers require that their PBX be protected by a firewall, so the user needs to check with their firewall manufacturer for compliance. The firewall must have the ability to 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 to 64,000 – must be opened (ALG) for audio
- Bandwidth uses multiple IPs to allow media from its gateways.
Bandwidth SIP trunks support certain attributes, such as DTMF, Dial Plans, Codecs, Signaling Protocol, and IP Protocol. These are outlined in greater detail below.
Dual-Tone Multi-Frequency (DTMF) signaling is used for detecting dialed digits over the SIP connection, either outbound or inbound. Bandwidth supports in-band or out-of-band DTMF outlined in RFC 2833.
Bandwidth supports only E.164 for outbound and inbound calling on our UC product.
This is the recognized international standard. It's characterized by a “+” followed by the country code (i.e. “1” for the U.S.) and then the specific phone number. Some phone systems don't have the ability to support this method. A user shouldn't choose the UC product unless they are sure of their PBX’s compliance. If this product is chosen, Bandwidth will be sending all calls into the PBX in E.164 format. We'll also expect that calls are sent back to us in this method.
- Example Local & Long Distance: +19192971100
- Example International: +4402074942020
Bandwidth requires that all SIP and Audio be delivered via UDP. TCP isn't currently supported. The UDP packets must be no larger than 1350 bytes. If a user sends UDP packets bigger than this, a Message Size Too Big error will keep the call from completing.
Bandwidth fully supports two codecs to date: G729a and G711ulaw. Inbound calls to Bandwidth numbers will also support ILBC.
Bandwidth SIP trunks utilize SIP features like Inbound Caller-ID, Outbound Caller-ID, Call Transferring, Conference Calling, and Forwarding. Please open a ticket with your Bandwidth Support Team if you have any questions about these supported features.
Caller ID In/Out
Bandwidth SIP uses the FROM field to represent the Caller-ID name, number, and call rating. If a Remote-Party-ID field (RPID) is included in the SIP INVITE message, the RPID will be used for Caller-ID and for call rating instead. The From field and the RPID should be in E.164 format.
Note: P-Asserted-Identity and Privacy headers are also supported.
CLID example headers
- From: "Bandwidth" <sip:+firstname.lastname@example.org>;tag=8dzbfe95km
- Remote-Party-ID: "Angela" <sip:+email@example.com>privacy=off;screen=no
- P-Asserted-Identity: "Raleigh NC" <sip:+firstname.lastname@example.org>
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