SIP Troubleshooting Checklists

SIP Connection Information

Property	Value in EU-Production	Value in EU-Staging	Value in US-Production
Inbound SIP-Proxy	sip.cognitivevoice.io	sip.demo.join-us.cloud	sip.atl.join-us.cloud
Inbound SIP-Proxy IP	145.14.231.149	145.14.231.171	204.124.15.179
Inbound SIP-Proxy Port	5060/UDP (plain), 5060/TCP (plain), 5061/TCP (TLS)	5060/UDP (plain), 5060/TCP (plain), 5061/TCP (TLS)	5060/UDP (plain), 5060/TCP (plain), 5061/TCP (TLS)
Outbound SIP-Proxy IP	145.14.231.153	145.14.231.175	204.124.15.183
RTP Port-Range (inclusive)	Start: 50000/UDP End: 57999/UDP	Start: 50000/UDP End: 57999/UDP	Start: 25000/UDP End: 64999/UDP

No whitelisting is necessary in CVG, any valid SIP communication is accepted and appropriate responses are sent out.

The CVG SIP proxies will generally answer to SIP OPTIONS ping.
Use a sensible User-Agent header because pings from commonly known scanner utilities will not be answered.
SIP OPTIONS pings must be properly addressed to the CVG endpoint. E.g. the Request-URI must refer to our inbound proxy, e.g. OPTIONS sip:sip.cognitivevoice.io ...

Call destinations must be international phone numbers with a leading country code in either +49... or 0049... formats. Other formats or amounts of leading zeros may lead to call rejection.
The request URI must refer to our inbound proxy, e.g.: INVITE sip:+4944444444@sip.cognitivevoice.io. Using the IP address and optionally an explicit SIP port (5060 or 5061 for TLS) is also fine.
The SIP URI in the From header must never refer to any of our SIP proxies. E.g. From: Someone <sip:+4944444444@sip.cognitivevoice.io> would not be valid since this would identify the user as a local user.
If no Privacy header is sent, then SIP addresses in the From header will be considered anonymous/supressed, if the display name is anonymous, the user-port of the URI is anonymous or the host ends with .invalid. All checks are done case-insensitive.

CVG does not accept REFER requests, however it is possible to REFER calls to CVG and it is possible to send out REFER requests from CVG using the /call/refer API.

Make sure the SIP communication works without encryption before enabling it!
TLS/SIP in combination with SRTP-SDES is supported using modern TLS 1.2 and TLS 1.3 cipher suites (this is usually used from contact center solutions).
TLS/WSS in combination with SRTP-DTLS is supported using modern TLS 1.2 and TLS 1.3 cipher suites (this is usually used for WebRTC).
Support for specific cipher suites can be probed using the curl command line tool via the URL https://sip.cognitivevoice.io:5061.

Advertised Codecs are: G.722, OPUS, G.711a and G.711u (prioritized in this order)