SIP Trunking lets businesses make and receive phone calls over the internet (VoIP) using the SIP standard. The global, geo-redundant CM.com SIP Trunking platform is split into inbound and outbound traffic with multiple Points-of-Presence.
Overview
Outbound calling
|
Region |
FQDN (preferred, DNS info) |
IP Addresses |
|---|---|---|
|
Europe |
nl.voip.cm.com |
31.169.63.1 / 31.169.63.2 / 31.169.62.162 / 31.169.62.163 |
|
Asia |
sg.voip.cm.com |
85.119.53.200 / 85.119.53.201 |
|
US |
us.voip.cm.com |
31.169.56.200 / 31.169.56.201 |
By default, when a call is routed to one of our regional PoPs, the media is also routed via that region. If a different PoP for media is desired, please contact [email protected].
It is advised to use DNS, only resort to using IP addresses when your system does not support DNS for signalling endpoints. CM.com supports DNS SRV and A records.
CM.com supports UDP (port 5060) and TLS (port 5061) for SIP signalling for outbound calling.
CM.com requires your IP addresses to be allow-listed and requires authentication. More information on authentication.
All numbers should be in E.164 format. The leading + can be omitted. CLIs should be valid. More information on number validation and restrictions.
Inbound calling
After purchasing or porting DIDs, CM.com will deliver inbound calls to your DID on predetermined endpoints. These endpoints can be managed via the Voice Management App (link). A Distribution group can have one or more IP or DNS endpoints listed. A phone number can be configured to route calls to these endpoints by selecting the desired Distribution group. The endpoints in the Distribution group will be tried either round-robin or fallback (in the order listed). Endpoints can be UDP or TLS endpoints. SIP REGISTER is not supported for inbound calling.
SIP OPTIONS messages are sent to each endpoint in a Distribution group. When an endpoint does not respond with a 200 OK, calls will not be routed to that endpoint until it does. When none of the endpoints in a Distribution group are responding accordingly, calls are not delivered and the number will be unreachable.
When using an ACL or firewall, please allow SIP signalling from the following CM.com IP subnets on the desired protocol (UDP, or TCP when using TLS) and port:
|
Region |
Subnet |
|---|---|
|
Europe (default) |
188.94.184.96/28
|
|
Asia |
85.119.53.128/25 |
|
US |
31.169.56.128/25 |
By default, CM.com delivers all calls from our EU region. If a different signalling PoP for inbound calling is desired, please contact [email protected].
By default, we route media for inbound calls via our NL PoP. If a different PoP for media is desired, please contact [email protected].
Media
CM.com uses the port range 10.000 - 60.000 (UDP) and the following subnets for media:
-
85.119.49.0/25
-
85.119.53.128/25
-
31.169.56.128/25
Please allow these subnets and ports in your firewall.
The following audio codecs are supported:
-
G.711 U (PCMU)
-
G.711 A (PCMA)
-
G.722 (best effort)
-
G.729
Other codecs might be accepted or offered by up-/downstream operators and CM.com transparently relays those codecs.
CM.com supports both DTLS-SRTP and SDES when encrypted media is desired. Encrypted media is only possible when signalling is encrypted using TLS. CM.com will automatically switch to DTLS-SRTP or SDES when offered on outbound calls. For media encryption on inbound calls, this can be configured per DID via the Voice Management App. More information on encryption.
Additional technical details
|
Signalling |
|
|---|---|
|
Protocols |
UDP and TLS |
|
Session timers |
Supported and strongly recommended |
|
Minimal session interval |
120s |
|
Maximum call duration |
3h |
|
MTU |
1500 bytes (recommended) |
|
Encryption |
|
|
Supported TLS versions |
1.2 and 1.3 |
|
RTP |
|
|
UDP port range |
10.000 - 60.000 |
|
Packetization |
20ms |
|
SRTP |
DTLS and SDES |
|
DTMF |
|
|
RFC 2833 / 4733 |
Supported |
|
SIP INFO |
Not supported |
|
Inband DTMF |
Not supported |
|
Fax |
|
|
Not supported |
|