Howto:FR - OpenIP - SIP Trunk Touch SIP-Provider (2016)

FIXME: This is a temporary version of innovaphone SIP Interop Tests. Merge back to official wiki when finished

FIXME: Article name must have format Country_-_Provider_-_Productname_-_SIP Provider (YYYY)

Summary

The tests for the SIP Trunk of the provider OpenIP were completed.

Issues found were:

Clip No Screening

Fax T38

SRTP

Redundancy Failover

Provider only offers 1 codec (G711U or G711A)

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 only UDP as transport protocol. As a result, the SIP-communication is not encrypted. Moreover it requires all involved network elements to support IP-fragmentation.

CLIP

OK

CLIR

OK

Clip No Screening(CLNS)

CLNS is not possible

However, redirection based on the Diversion number or based on "302 Moved Temporary" is not possible. As a result, forwarded calls or Mobility calls will not see the original calling-number but the diverting parting number as caller(if the TrunkLine - object is configured with Set Calling=Diverting No)

Codecs

The provider support the following codecs: G711u, G711a (However only one can be used at time, it's necessary to ask to switch codec).

Fax

Transport of faxes to/from the PSTN via G.711(A/U) codec was tested successfully. However transport of faxes to/from the PSTN using the T.38 protocol was not tested successfully. This is important for the innovaphone Fax-server. Since the provider doesn't support T.38, you must use the "Audiofax" feature(i.e 2 DSPs) for each fax-call done by the innovaphone Fax-interface.

SRTP

The provider does not support audio encryption using SRTP.

DTMF (RFC2833)

OK

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.

Reverse Media Negotiation

OK

Mobility Call

Transmitting DTMF-tones as SIP-INFO messages was not possible, however we can do Mobility-calls since we use MediaRelay on the SIP-Interface.

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

Ringing not signalled by the provider. This will lead to incorrect call-state display on the PBX (phone-UI, myPBX, Soap) for outbound calls to the PSTN. The caller will see no status-update on the phone-display/PC-screen, showing that the remote party was reached and is ringing.
Additionally external callers forwarded/transferred back to the PSTN, will get no ring-tone but hear silence while the remote party is ringing. This silence while waiting might lead to aborting the call.

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

Listed here are only the test-results that differ from the tests with MediaRelay.

Mobility Call

Transmitting DTMF-tones as SIP-INFO messages was not possible, which is required in-order to use Mobility-calls without MediaRelay on the SIP-Interface.

Configuration

• Use profile Profile-Name in the Gateway/Interfaces/SIP menu.

Additional Configuration

• if no point below applies, don't use the Header ===Additional Configuration===
• check WORKING_AUDIOFAX_PSTN_REQUIRES_EXCLUSIVE_CODEC

if yes, check if G.711A is in WORKING_CODECS use text: FAX requires exclusive codec. Enable G711A Exclusive on all Fax-endpoints.

if yes but only G.711U is in WORKING_CODECS use text: FAX requires exclusive codec. Enable G711U Exclusive on all Fax-endpoints.

• if WORKING_NAT_SCHEMES contains D but no C, then the provider can work without MR but requires STUN on all endpoints

use text: If MediaRelay is not enabled on the SIP-Trunk, all RTP-endpoints must have a STUN-server configured.