Howto:SIP Interop Test Description
innovaphone SIP Interop Tests
Summary
This article describes the SIP provider Tests done by innovaphone when testing the compatibility of a provider. The features are divided in two classes: KO-criteria and optional criteria. When the provider does not pass a test marked as KO-criteria, the provider can not be accepted in the category recommended SIP Provider.
To differentiate the quality of the offered service, the provider receives an additional rating. The current rating value ranges between 0 and 100 points. The test success rate is displayed as a percent value. It is calculated by using the following formula: (rating points achieved / rating points total) * 100.
For test results, see the list of tested SIP providers.
Test 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.
Basic Call
call using g711a(KO)
Purpose: Test of capability to handle G711A RTP - Streams
Test description:
- Tel1 is configured with G711A exclusive coder preference.
- Tel1 calls a phone in the PSTN
call using g711u(KO)
Purpose: Test of capability to handle G711U RTP - Streams
Test description:
- Tel1 is configured with G711U exclusive coder preference.
- Tel1 calls a phone in the PSTN
call using g723(optional)
Purpose: Test of capability to handle G723 RTP - Streams
Test description:
- Tel1 is configured with G723 exclusive coder preference.
- Tel1 calls a phone in the PSTN
call using g729(optional)
Purpose: Test of capability to handle G729 RTP - Streams
Test description:
- Tel1 is configured with G729 exclusive coder preference.
- Tel1 calls a phone in the PSTN
Overlapped sending(optional)
Purpose: Test if the provider supports Overlap Dialing. If not Enblock Dialing must be configured on the SIP - trunk.
Test description:
- Route from PBX to SIP-provider is not configured with 'Force Enblock'.
- Tel1 calls a phone in the PSTN
early media channel(KO)
Purpose: Test if the transmission of RTP - packets is possible before the call has been accepted (200 OK) by both endpoints. This feature is used when the caller receives announcements or dialtones from the SIP - Provider. (i.e. 'The dialed number is incomplete.')
Test description:
- Tel1 calls a not existant(incomplete) phonenumber in the PSTN.
- PSTN-Provider will play an announce, i.e. 'Number incomplete'.
- The SIP-provider forwards the announcement and terminates the call setup procedure.
Fax using T.38(optional)
Purpose: Test of capability to handle T.38 signalling/data. Without this feature Fax over IP is not possible.
Test description:
- Fax1 sends a message to a fax machine in the PSTN.
- Fax1 receives a message from a fax machine in the PSTN.
CGPN can be supressed(optional)
Purpose: Test if provider accepts anonymous calls
Test description:
- Tel1 is configured to not send his CGPN. 'User Setup'->'Number Presentation' = Off
- Tel1 calls a phone in the PSTN. The display of the PSTN phone should show as Calling Party: 'anonymous'
Reverse Media Negotiaton(KO)
Purpose: Test if Reverse Media Negotiation is implemented by the provider.
Test description:
- Tel1 calls Tel2. Tel2 picks up the call.
- Tel2 makes a blind transfer to a phone in the PSTN.
- This is the quickest way to send a INVITE message without SDP over the SIP-Trunk.
Voice Quality OK?(KO)
Purpose: Simple test of overall voice quality during a call.
Test description:
- Tel1 calls Tel2. Tel2 picks up the call.
- Manual test of voice quality by speaking/listening on both ends.
Direct Dial In
Inbound(Provider -> Innovaphone)(KO)
Purpose: Test if the provider forwards the call to the PBX using the correct DDI - number(CDPN).
Test description:
- Tel1 calls phone in the PSTN.
- Check if the PSTN phone display shows the correct CGPN.
Outbound(Innovaphone -> Provider)(KO)
Purpose: Test if the provider handles the CGPN (trunk number + extension) correctly.
Test description:
- Phone in the PSTN calls Tel1.
- Check if the CDPN is forwarded correctly to the PBX.
DTMF
DTMF is also a must have feature for a company. DTMF is crucial for the use of a voicemail system. Currently there are two methods of transfering DTMF signals, by SIP - INFO message or encapsulated in the RTP - packet. Innovaphone supports both types of DTMF signalling. However you must pay attention at the proper configuration of your innovaphone box, since your provider will typical support just one kind of DTMF tone siganlisation.
DTMF tones sent correctly(KO)
Purpose: Test if DTMF signals going from PBX to Provider are received and interpreted correctly.
Test description:
- Tel1 calls phone in the PSTN.
- Check if the DTMF - tones are received at the PSTN phone(using SOAP).
DTMF tones received correctly(KO)
Purpose: Test if DTMF signals going from Provider to PBX are received and interpreted correctly.
Test description:
- PSTN Phone calls Tel1
- Check if the DTMF - tones are received at Tel1(using SOAP).
Hold/Retrieve
When a call is put on hold, users normally expect to hear some kind of music/announcment signalling them that they should wait. However there are two possibilities. The PBX generates the announcement or the provide generates it.
To test a PBX generated announcement, use the R - key to hold a conversation. This type of holding is tested in the Hold/Retrieve , Transfer with consultation and Transfer with consultation (alerting only) scenario.
To test a provider generated announcement, use the redial key to hold the conversation. This is used when doing a blind transfer.
Call can be put on hold(KO)
Purpose: Test if provider handles hold signalisation by Reinvite correctly.
Test description:
- Tel1 calls phone in the PSTN.
- Tel1 presses the 'R-Key'. Test if call is on hold(display blinking & MoH/announcement).
- Tel1 presses again the 'R-Key'. Test if call is retreived and conversation is continuable.
Held end hears music on hold / announcement from PBX(optional)
Purpose: Test if provider handles hold using the sendonly attribute correctly. The MoH will be transmitted by the PBX to the provider and must be then forwarded to the waiting phone.
Test description:
- Tel1 calls phone in the PSTN.
- Tel1 presses the 'R-Key'. Test if call signalisation is correct and MoH is audible on waiting phone.
Held end hears music on hold / announcement from provider(optional)
Purpose: Test if provider handles hold using the inactive attribute correctly. The MoH will be transmitted by the Provider to the PBX and must be then forwarded to the waiting phone.
Test description:
- Tel1 calls phone in the PSTN.
- Tel1 presses the 'Redial-Key' and dials number of Tel2.
- Test if call signalisation is correct and MoH is audible on waiting phone.
Transfer with consultation
Call can be transfered(KO)
Purpose: Test of call-transfer with consultation
Test description:
- Tel1 calls phone in the PSTN.
- Tel1 presses the 'R-Key' and dials the number of Tel2.
- Test if audio channels between Tel1 and Tel2 are established correctly.
- Tel1 hangs up. PSTN phone changes its status from 'hold' to 'active'.
- Test if audio channels between PSTN phone and Tel2 are established correctly.
Held end hears music on hold(optional)
Purpose: Test for MoH/Announcemnet on hold phone.
Test description:
- Tel1 calls phone in the PSTN.
- Tel1 presses the 'R-Key' and dials the number of Tel2.
- Test if PSTN phone hears MoH.
Transfer with consultation (alerting only)
Call can be transfered(KO)
Purpose: Test of call-transfer with consultation
Test description:
- Tel1 calls phone in the PSTN.
- Tel1 presses the 'R-Key' and dials the number of Tel2.
- Tel1 doesn't wait for Tel2 to pickup the call and hangs up. PSTN phone changes its status from 'hold' to 'active'.
- Test if audio channels between PSTN phone and Tel2 are established correctly.
Held end hears music on hold(optional)
Purpose: Test for MoH/Announcemnet on hold phone.
Test description:
- Tel1 calls phone in the PSTN.
- Tel1 presses the 'R-Key' and dials the number of Tel2.
- Test if PSTN phone hears MoH.
Blind Transfer
Call can be transfered(optional)
Purpose: Test of call-transfer with consultation
Test description:
- Tel1 calls phone in the PSTN.
- Tel1 presses the 'Redial-Key' and dials the number of Tel2.
- Call is passed to Tel2 and Tel2 picks up the call.
- Test if audio channels between PSTN phone and Tel2 are established correctly.
Held end hears music on hold(optional)
Purpose: Test for Announcemnet/dial tone on hold phone.
Test description:
- Tel1 calls phone in the PSTN.
- Tel1 presses the 'Redial-Key' and dials the number of Tel2.
- Test if PSTN phone hears an announcement from the SIP - provider
Broadcast Group & Waiting Queue
Caller can make a call to a Broadcast Group(KO)
Purpose: Test of basic funtionality of the Broadcast feature, using Reinvite.
Test description:
- PSTN Phone calls a 'Broadcast Group' number.
- Tel1 and Tel2 are ringing. Tel1 picks up the call.
- Test if audio channels between PSTN phone and Tel1 are established correctly.
Caller can make a call to a Waiting Queue(KO)
Purpose: Test of basic funtionality of the WaitingQueue feature, using Reinvite.
Test description:
- PSTN Phone calls a 'Waiting Queue' number.
- Tel1 and Tel2 are ringing. Tel1 picks up the call.
- Test if audio channels between PSTN phone and Tel1 are established correctly.
Announcement if nobody picks up the call(KO)
Purpose: Test if announcement feature at Waitingqueues is working correctly.
Test description:
- PSTN Phone calls a 'Waiting Queue' number.
- Tel1 and Tel2 are ringing.
- PSTN phone hears an announcement from the PBX, i.e. 'All operators are busy. Please hold the line.'
- Tel1 picks up the call.
- Test if audio channels between PSTN phone and Tel1 are established correctly.