Howto:LU - MIXvoip - SIP Trunk SIP-Provider (2021)

Summary

The tests for the Mixvoip SIP Trunk of the provider Mixvoip were completed successfully.

Issues found were:

For more details about this issues, see the respective test-results sections.

Current test state

This product is being tested right now. The test is not yet completed.

Tests with MediaRelay

Registration

The provider supports UDP and TCP as transport protocol. The tests were completed using TCP, since UDP is an unreliable protocol and requires all involved network elements to support IP-fragmentation.

CLIP

OK

CLIR

CLIR didn't work

Clip No Screening(CLNS)

CLNS is not possible, however

• Redirection using "302 Moved Temporary" is possible. In case of a call forward, the incoming call is redirected to the PSTN. From point of view of the PBX the call doesn't exist after the redirection and is handled by the SIP-Provider. This is equivalent to "Partial Rerouting/Call Deflection" in ISDN. When a redirection is done, the redirection target will receive the original calling number, so for forwarded calls this will have the same effect as CLNS. However this redirection is not usable for Mobility calls or plain CLNS-calls. A plain CLNS-call is an outbound call with a "wrong" CGPN number (i.e. one that does not belong to the trunk). This happens e.g. if a call-center wants to send a service number (such 0800xxx) as CLI
• CLNS based on Diversion-Information is possible. Mobility calls and forwarded calls will show the original CGPN on the outbound call. However this redirection is not usable for plain CLNS-calls. A plain CLNS-call is an outbound call with a "wrong" CGPN number (i.e. one that does not belong to the trunk). This happens e.g. if a call center wants to send a service number (such 0800xxx) as CLI

Codecs

The provider support the following codecs: G711A, G711U

Fax

testing not yet concluded

SRTP

The provider does not support audio encryption using SRTP

DTMF (RFC2833)

Transmitting DTMF-tones as RTP-events was not possible. This may create interoperability issues with non-innovaphone devices

NAT Traversal

The provider detects clients behind NAT and can handle calls to them without requiring the clients to use NAT-traversal methods like STUN. However MediaRelay is required, since the provider does not support a change of the remote RTP-endpoint during a call

Reverse Media Negotiation

OK

Mobility Call

OK

Redundancy

Registration of two SIP-interfaces on the same SIP-account is supported by the provider. However, the provider has no failover mechanism if one device is down. As a result, you can use both SIP-interfaces for load-balancing purposes. If one device is down, for a duration of up to 2 minutes (i.e registration interval) incoming and outgoing calls might be rejected/fail.

IP-Fragmentation

OK

Large SIP messages

OK

Correct signalling of Ringing-state

OK

Early-Media

The provider supports early-media for outbound calls to the PSTN

Session Timer

The tests regarding the SIP-session timer were successful. This means that either no session expiry is used or that it is used and works. It does not imply that session expiry actually is used

Call Transfer

OK

Tests without MediaRelay

The tests without MediaRelay were aborted, since it is required by the provider. The reason for it, are audio problems when changing the remote RTP-endpoint

Configuration

• Use profile LU-MIXvoip-SIP_Trunk in the Gateway/Interfaces/SIP menu.

Additional Configuration

The following settings are not done by the configuration profile and must therefore be done manually:

• FAX requires exclusive codec. Enable G711A Exclusive on all Fax-endpoints