Howto:FR - OVH Telecom - SIP Trunk SIP-Provider (2016)
Jump to navigation
Jump to search
Summary
The tests for the SIP-trunk of the provider OVH Telecom were completed successfully.
Issues found were:
- Clip No Screening
- Fax
- SRTP
- Mobility Call
- Redundancy
- Correct signalling of Ringing-state
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 May 24th, 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 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.
- To use call redirection with 302 Moved temporary, you have to enable Reroute supported at your TrunkLine PBX-object. Additionally you must enable Interworking on the inbound route from the Provider to the PBX(i.e. SIPx -> RSx).
- Codecs
- The provider support the following codecs: G711A, G711U, G729A. The following codecs are not supported: G722, Opus.
- 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
- OK
Configuration
- Use profile (e.g. FR-OVH_Telecom-SIP_Trunk) in the Gateway/Interfaces/SIP menu.