Howto:SilverServer SIP Provider Compatibility Test
Innovaphone Compatibility Test Report
The tests for this product have been completed.
Summary
SIP Provider: SilverServer(Austria)
The provider supports only G711 as audio codec, also T.38 is not supported. However reverse media negotiation(needed for blind transfer) and DDI are supported.
The configuration on innovaphone side is difficult, since you must use the VoIP-GW to connect to the provider. More on this topic in chapter SilverServer SIP Compatibility Test
SilverServer does not support the hold feature correctly. When making a transfer the remote end will not hear a MOH/dialtone.
Due to the unsupported hold feature, innovaphone does not reccommend to use this provider.
- Features:
- Direct Dial In
- DTMF
- Supported Codecs by the provider
- G711
Testing Enviroment
Scenario
This scenario tries to be as simple as possible. The IP - PBX and the phones are all in the same network, having public IP - adresses. The signalling channel will pass through the PBX, the media channel will go from endpoint (phone) to endpoint (provider).
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 supressed | No |
| Reverse Media Negotiaton | 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 received correctly | Yes |
Hold/Retrieve
| Tested feature | Result |
|---|---|
| Call can be put on hold | Yes |
| Held end hears music on hold / announcement from PBX | No |
| Held end hears music on hold / announcement from provider | No |
Transfer with consultation
| Tested feature | Result |
|---|---|
| Call can be transfered | Yes |
| Held end hears music on hold | No |
| Call returns to transferring device if the third
Endpoint is not available |
Yes |
Transfer with consultation (alerting only)
| Tested feature | Result |
|---|---|
| Call can be transfered | Yes |
| Held end hears music on hold or dialing tone | No |
| Call returns to transferring device if the third
Endpoint is not available |
Yes |
Blind Transfer
| Tested feature | Result |
|---|---|
| Call can be transfered | Yes |
| Held end hears dialing tone | No |
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: 6.00 dvl-sr1 IP800[07-60600.74]
- IP22: 6.00 sr1-hotfix4 IP22[07-60600.72]
- IP200: 6.00 dvl-sr1 IP230[07-60600.58]
- IP230: 6.00 dvl-sr1 IP230[07-60600.58]
SIP - Trunk
First of all the SIP Trunk must be configured. Since SilverServer authenticates a user account only by IP - Address, you cannot use the normal SIP-Gateway object. You need to configure a GW without registration and route the calls to the PBX. Also SilverServer uses different servers for load balancing reasons. You must make an GW without registration object for every possible IP-destination(e.g. SilverServer SIP Server).
Like described, before you need a Gateway for every possible SilverServer SIP Server. Note that the Server Adrress on the two GWs changes, while the Domain entry does not. Another important setting is to change the SIP Interop Tweak from AOR to AOR with CGPN as display on both GWs(i.e. GW1 & GW3). You have to do this, because SilverServer does not read the Preffered identity header information but looks directly in the FROM header for DDI information.
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 blockwise sending of the phone number. You can do this by enabling Force enblock in your routes.
The second setting you must check is Interworking(QSIG,SIP). This feature must be enabled to properly relay suplementary services, like Hold over the SIP Trunk. If this checkbox is unchecked only basic call Information like connect and disconnect will be forwarded by the Gateway.
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.