Howto:FR - OpenIP - SIP Trunk Touch SIP-Provider (2016): Difference between revisions
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== Tests without MediaRelay== | == Tests without MediaRelay== | ||
Listed here are only the test-results that differ from the tests with 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. | |||
<!-- only needed for tests without nightly-test execution | <!-- only needed for tests without nightly-test execution |
Revision as of 11:35, 10 June 2016
FIXME: This is a temporary version of innovaphone SIP Interop Tests. Merge back to official wiki when finished |
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.