General SIP Information
SIP Signaling Addresses
DIDWW may originate inbound calls to your SIP endpoint using two delivery methods:
Static SIP URI
SIP Registration
Each method uses different SIP endpoints.
Static SIP URI
When forwarding calls to your equipment using a static SIP URI, calls will originate from the following DIDWW SIP endpoints:
Location |
IPv4 address |
IPv6 address |
|---|---|---|
New York |
46.19.209.14 |
2a01:ad00:1:14::14 |
Frankfurt |
46.19.210.14 |
2a01:ad00:2:14::14 |
Los Angeles |
46.19.212.14 |
2a01:ad00:4:14::14 |
Miami |
46.19.213.14 |
2a01:ad00:5:14::14 |
Singapore |
46.19.214.14 |
2a01:ad00:6:14::14 |
Hong Kong |
46.19.215.14 |
2a01:ad00:7:14::14 |
Amsterdam |
185.238.173.14 |
2a01:ad00:8:14::14 |
SIP Registration
When using SIP Registration to dynamically register your SIP trunk, you must register to the following DIDWW SIP endpoint:
Host |
IPv4 address |
IPv6 address |
|---|---|---|
sip.didww.com |
185.238.173.49 |
2a01:ad00:8:1::49 |
SIP Signaling Source Ports
DIDWW originates inbound SIP signaling using the following source ports, regardless of whether SIP URI delivery or SIP Registration is used:
UDP: source port
5060TCP / TLS: dynamic source ports
5070–65535
Ensure that your firewall or SBC allows SIP signaling from the DIDWW signaling IP addresses listed in the sections above.
RTP addresses
Our system sends RTP packets from the following subnets:
46.19.208.0/21
185.238.172.0/22
2a01:ad00:1:2::/64
2a01:ad00:1:14::/64
2a01:ad00:2:1::/64
2a01:ad00:2:14::/64
2a01:ad00:4::/64
2a01:ad00:4:14::/64
2a01:ad00:5::/64
2a01:ad00:5:14::/64
2a01:ad00:6::/64
2a01:ad00:6:14::/64
2a01:ad00:7:1::/64
2a01:ad00:7:14::/64
2a01:ad00:8:1::/64
2a01:ad00:8:14::/64
with RTP port-range: 1024-65535
RTCP
We transmit and receive RTCP packets from port = rtp_port + 1 (as recommended in RFC3550 ). Additionally, we are able to receive RTCP packets on the same port as RTP, considering utilizing RTCP conflicts avoidance payloads (payload types 72-76).
SIP OPTIONS
DIDWW is responding on SIP OPTIONS while sending requests to signaling IPs. It will respond on UDP 5060, TCP 5060 and TLS 5061 ports.
Please note that SIP OPTIONS responses are not related to call processing.
Encryption
Our systems supports TLS as a secure transport for SIP signaling and SRTP as a media encryption mechanism. All three SRTP key negotiation mechanisms are supported - SDES , DTLS and ZRTP .
Warning
The SRTP media encryption and TLS SIP transport are disabled by default. Please contact our sales team sales@didww.com to allow traffic encryption for your account.
Supported codecs
G.711 A-law/U-law
G.729
G.723.1
L16
G.726-16/G.726-40/G.726-32/G.726-24
G.721
GSM
Speex
DTMF transport methods
DTMF signaling is supported as follows:
Telephone-event RFC2833
SIP INFO draft-kaplan-dispatch-info-dtmf-package-00
application/dtmf-relay
application/dtmf
By default RFC 2833 enabled