Howto:AT - Russmedia IT - highspeed Telefon SIP-Provider (2016)
Jump to navigation
Jump to search
Summary
The tests for the highspeed Telefon of the provider Russmedia IT GmbH were completed successfully.
Issues found were:
- SRTP
- NAT Traversal
- Mobility Call
- Redundancy
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 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)
- OK
- Codecs
- The provider support the following codecs: G711A The following codecs are not supported: G711U, G729, G722, Opus.
- Fax
- Transport of faxes to/from the PSTN via G.711(A/U) codec was tested successfully. Additionally transport of faxes to/from the PSTN using the T.38 protocol was tested successfully. This is important for the innovaphone Fax-server. Even if the provider supports T.38, it is not guaranteed that all Fax-calls use T.38. However each call using T.38 will save you 2 DSP-licenses on the gateway hosting the 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. However MediaRelay is required, since the provider expects that all RTP-packets are passed through the PBX.
- Reverse Media Negotiation
- OK
- 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.
- 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
- OK
- 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 two external RTP-endpoints are connected(e.g. external transfer, mobility call).
Configuration
- Use profile (e.g. AT-Russmedia_IT-highspeed_Telefon) in the Gateway/Interfaces/SIP menu.