TABLE OF CONTENTS



Incoming (Dial-In) 


This configuration is for dial-in, that is, calls from a telephone in the PSTN in to the conferencing service/platform


What follows are basic configuration requirements to support interoperability with iotum.


Description

IP Address(es)

Ports

SIP Signaling (Primary)

161.35.253.199 

UDP: 5060

SIP Signaling (Secondary)

64.225.91.54

UDP: 5060

RTP Media

(see list below)

UDP: 1025-32768



DTMF RFC2833 Only

Caller ID format:  E.164 (ie 443307771307)

Please use E.164 format, even for NANPA numbers, in the To: of SIP INVITES, e.g. To: <sip:14165551212@137.184.201.74:5060>


Supported Codecs:  G.722 (preferred), G.711U/G.711A


Required information from you for iotum to finalize configuration:

*email us at operations@iotum.com and pls cc cara@iotum.com ) 

  1. The signaling IPs on your side, so we may whitelist them in our infrastructure;

  2. The numbers you will be using in INVITEs’ To: so we may associate them with your white label instance.


Important notes about our signaling IPs

The values above are typical; however, select the one with the lowest latency for your primary, and the second-lowest for your secondary, from the following list:


161.35.253.199 (US east coast)

64.225.91.54 (US west coast)

143.244.204.70 (Frankfurt)

139.59.193.50 (Singapore)

139.59.48.42 (Bangalore)


RTP can be any of:

52.64.204.228

64.68.167.102

64.68.167.124

64.68.184.146

64.68.184.150

67.55.209.108

67.55.209.109

67.55.209.121

67.55.209.122

137.184.138.241

137.184.193.40

137.184.201.74

137.184.205.245

143.198.184.75

147.182.143.197

147.182.143.197

147.182.162.238

147.182.162.238

147.182.202.128

159.223.25.122

159.223.25.179

159.223.109.211

159.223.149.36

159.223.153.135

159.223.155.152

159.223.157.16

164.92.64.145

164.92.64.148

165.232.130.174

165.232.130.174

165.232.188.144

165.232.188.152

188.166.253.84

206.189.61.173

206.189.61.173

216.249.240.21

216.249.240.22

216.249.240.23

216.249.240.24

216.249.240.25

216.249.240.26







Outgoing (Dial-Out)


This configuration is for dial-out, that is, calls from the conferencing service/platform out to a telephone in the PSTN. This is less common than Incoming/Dial-in (for which see its separate documentation).


What follows are basic configuration requirements to support interoperability with iotum.


Description

IP Address(es)

Ports

SIP Signaling

(see list below)

UDP: 5060

RTP Media

UDP: 1025-32768



DTMF RFC2833 Only

We will send From: and To: in INVITES as full E164 including the leading + (i.e. +443307771307)

Supported Codecs:  G.722 (preferred), G.711U/G.711A

To avoid toll fraud and related issues, please configure trunks to either or both:

  • accept INVITES from us only from any of the IP addresses below;

  • require digest authentication (and, of course, generate and provide us credentials)




Please supply us with IP address[es] or FQDN[s] of outbound proxy[ies] and any special instructions about the order in which to try them, etc.




SIP Signaling and RTP can be any of:

52.64.204.228

64.68.167.102

64.68.167.124

67.55.209.108

67.55.209.109

67.55.209.121

67.55.209.122

137.184.138.241

137.184.193.40

137.184.201.74

137.184.205.245

143.198.184.75

147.182.143.197

147.182.162.238

147.182.202.128

159.223.25.122

159.223.25.179

164.92.64.145

164.92.64.148

165.232.130.174

165.232.188.144

165.232.188.152

188.166.253.84

206.189.61.173

216.249.240.21

216.249.240.22

216.249.240.23

216.249.240.24

216.249.240.25

216.249.240.26