Howto:SilverServer SIP Provider Compatibility Test
Innovaphone Compatibility Test Report
Summary
SIP Provider: SilverServer(Austria)
- Features:
- Direct Dial In
- DTMF
- Supported Codecs by the provider
- G711
The provider does support all required innovaphone features and is therefore qualified as recommended SIP Provider.
SilverServer has achieved 85% of all possible test points. For more information on the test rating, please refer to Test Description
Current test state
The tests for this product have been completed.
Testing of this product has been finalized June 20th, 2011.
Testing Enviroment
Scenario
This scenario describes a setup where the PBX and phones are in a private network. The IP800 must use a stun server, in order to send correct SIP - messages. The IP800 works as media relay, all RTP - streams go through the PBX.
Test Results
For more information on the test procedure, please read the following wiki article: SIP Interop Test Description. Bold lines in the test results indicate a KO-criteria.
Basic Call
Tested feature | Result |
---|---|
call using g711a | Yes |
call using g711u | Yes |
call using g723 | No |
call using g729 | No |
Overlapped sending | No |
early media channel | Yes |
Fax using T.38 | No |
CGPN can be suppressed | No |
CLIP no screening | No |
Reverse Media Negotiation | Yes |
Long time call possible | Yes |
External Transfer | Yes |
Voice Quality OK? | Yes |
Direct Dial In
Tested feature | Result |
---|---|
Inbound(Provider -> Innovaphone) | Yes |
Outbound(Innovaphone -> Provider) | Yes |
DTMF
Tested feature | Result |
---|---|
DTMF tones sent correctly | Yes |
DTMF tones sent correctly via SIP-Info | Yes |
DTMF tones received correctly | Yes |
Hold/Retrieve
Tested feature | Result |
---|---|
Call can be put on hold | Yes |
Held end hears music on hold / announcement from PBX | Yes |
Held end hears music on hold / announcement from provider | Yes |
Transfer with consultation
Tested feature | Result |
---|---|
Call can be transferred | Yes |
Held end hears music on hold | Yes |
Call returns to transferring device if the third Endpoint is not available | Yes |
Transfer with consultation (alerting only)
Tested feature | Result |
---|---|
Call can be transferred | Yes |
Held end hears music on hold or dialling tone | Yes |
Call returns to transferring device if the third Endpoint is not available | Yes |
Blind Transfer
Tested feature | Result |
---|---|
Call can be transferred | Yes |
Held end hears dialling tone | Yes |
Broadcast Group & Waiting Queue
Tested feature | Result |
---|---|
Caller can make a call to a Broadcast Group | Yes |
Caller can make a call to a Waiting Queue | Yes |
Announcement if nobody picks up the call | Yes |
Configuration
General Information
Firmware version
- IP800: 9.00 hotfix1
- IP22: 8.00 hotfix16 - build 805005400
- IP200: 9.00 hotfix1
- IP230: 9.00 hotfix1
SIP - Trunk
First of all the SIP Trunk must be configured. Since SilverServer authenticates a user account by IP - Address and by Registration, you mus use the normal SIP-Interface and a Gateway - Interface without registration.
Outgoing calls to the SIP provider must be sent via the SIP - Interface
A important setting is to change the SIP Interop Tweak 'From Header when Sending INVITE' to CGPN in user part of URI and Identity Header when Sending INVITE to 'Fixed AOR'. You have to do this, because SilverServer does not read the Preferred identity header information but looks directly in the FROM header for DDI information.
Since no NAT - detection is performed by the provider, 'Media-Relay' must be activated and a STUN server entry must be made, when the PBX is in a private IP - network - behind a NAT router. The NAT router must be STUN capable.
The GW without registration is needed to route the calls to the PBX.
Again, the interface settings are done similar to the SIP interface. Most important, 'Media-Relay' must be activated and a STUN server entry must be made, when the PBX is in a private IP - network - behind a NAT router. The Gateway interface must have the 'Local Domain' configured with the IP - address of the customer NAT - router.
The calls are sent and received from different servers on Silverserver site. As a result, you will have a different entry as 'AOR'/'Remote Domain' on the SIP interface compared to the Gateway interface.
Number Mapping
The complicated part on this issue is the correct mapping of the outgoing and incoming numbers.
Route Settings
Because SilverServer, as most SIP - Providers too, doesn't support overlap sending, you must enable the block-wise sending of the phone number. You can do this by enabling Force enblock in your routes.
The second setting you can check is 'Interworking(QSIG,SIP)'. When enabling this feature, you will hear the MoH offered by the provider. If 'Interworking(QSIG,SIP)' is disabled, a 'Call Hold' will not be forwarded to the provider. As a result, you will hear in this case the MoH provided by the PBX. It is recommended to leave the 'Interworking(QSIG,SIP)' unchecked, as shown in the screenshot below.
Now the PBX and the phones are setup correctly. You should be able to make call in both directions and send and receive fax messages.