Howto:FR - OpenIP - SIP Trunk Touch SIP-Provider (2016)
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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
The tests for this product have been completed. See the Summary section for more details.
Testing of this product has been finalized June 10th, 2016.
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 FR-OpenIP-SIP_Trunk in the Gateway/Interfaces/SIP menu.
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