Reference12r2:Phone/User/General

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There are other versions of this article: Reference | Reference9 | Reference10 | Reference11r1 | Reference12r1 | Reference12r2 (this version)

Here, you enter the parameters for registering with the innovaphone PBX. These parameters are preset by the administrator of the telephone system. First, you must select the required protocol for registration. The options are H.323 or SIP. The respective parameters are automatically adjusted to the selected protocol.


Contents

H.323 protocol

Protocol
H.323
Basic H.323 protocol with RAS using UDP and call signaling using TCP
H.323/TCP
H.323 with tunneling RAS over the signaling (H.460.17), so that only a single TCP connection is used
H.323/TLS
H.323 with tunneling RAS over the signaling (H.460.17) and signaling using TLS. If this option is together with registration by hardware id, the innovaphone device certificate can be used for authentication.
Primary Gatekeeper Address
Here, you specify the DNS Name or IP address at which the first responsible gatekeeper can be reached.
Secondary Gatekeeper Address
There should be a further gatekeeper in the network in case the first gatekeeper is unavailable. You enter DNS Name or the IP address of this gatekeeper here.
NOTE: When the phone works with Gatekeeper Discovery (Gatekeeper identifier) as primary registration and a Secondary gatekeeper address is configured, this direct IP address will be used as backup. So when the Gatekeeper discovery fails the secondary gatekeeper address is used to register.
Gatekeeper Identifier
If several gatekeepers are to be active at one address, a particular gatekeeper is identified amongst them using the name entered here.
NOTE: If the phone is registered at an innovaphone PBX and the phone is configured on a different PBX then the local PBX, thus the registration is redirected, a Gatekeeper Identifier of <local-pbx>@<System Name> should be used. This way the mechanisms asscociated with this (calling of local objects) will work even if the registration was not redirected.
Gatekeeper Certificate
For H.323/TLS registrations. The name of the certificate expected to be used by the gatekeeper. If configured this name is validated when the TLS session is set up. Useful to verify on the phone that the registrations was accepted by the correct Gatekeeper/PBX.
Number
Here, you specify the call number required for the registration. This number is only required if no name was specified.
Name
The name you enter here is only required for the registration if the number was not specified.
Local endpoint address
If several local IP addresses are available on the IP telephone, this setting enables you to specify the one to be used for communication with the gatekeeper.
Note: Since the local IP address was preset for the IP telephone using the IP routing table, the value is only necessary in exceptional cases.
Also we can define a fixed port for the H.323 Signalling. By default the User-1 tab uses port 1720 and all other users use a random port. In some scenarios were NAT/Firewall Routers block/intercept port 1720 we can define a different port for the User-1 by setting ":Port" ex: :1920.
Note:If we set fixed port for each Registration on the IP Phone we must be careful to not use the same port twice to avoid any conflicts and make sure this port it's not blocked by any firewall

SIP protocol

Protocol
SIP/UDP
SIP signaling using UDP as transport protocol (RFC 3261).
SIP/TCP
SIP signaling using TCP as transport protocol (RFC 3261).
SIP/TLS
SIP signaling using TLS as transport protocol (RFC 3261).
Primary server address
IP address at which the responsible SIP server may be reached.
Secondary server address
There should be a further SIP server in the network in case the first one is unavailable.
Note: Since the local IP address was preset for the IP telephone using the IP routing table, the value is only necessary in exceptional cases.
Domain
Instead of the IP address, you can specify the domain of the provider, which is found after the @ of the URI.
Remote Certificate
In case the domain name of the remote certificate differs from the domain of the provider, alternate domain name in the remote certificate can be configured here.
CGPN
The calling party number to be used on outgoing calls.
User ID
The user ID, which corresponds to the part in front of the @ of the URI.
Note: The user ID can be numeric or alphanumeric.

Protocol independent

The following parameters can be set protocol independently. These parameters are automatically negotiated during connection setup. A change is only necessary if the gatekeeper/server has special requirements regarding the connection protocol.

Username
In some SIP registrations, a separate user name is required for authorization. In all other cases, the field should be left empty.
Password
The registration requires a password, which can be agreed in this setting.
Retype
The confirmation prompt for the password.
Dial tones
In some countries, the dial tones are different. Here, you make the pre-settings for the specific country.
Enblock dialling timeout [s]
The IP telephone supports single digit dialling (overlapped sending), i.e. after a setup request the digits are sent as typed. After all digits of the number have been sent the connection is established automatically by the exchange. If the exchange does not support single digit dialling, you can switch to en-bloc dialling mode by entering a timeout value here. In en-bloc dialling mode the typed digits are gathered locally. With each digit the timeout is started. If there is no digit entered in the timout period a setup request containing the gathered digits is sent.
Note: Especially the IP150 phone does not allow entering the number without pick-up the headset. So it makes sense to use this timeout for enblock dialing at the IP150.
STUN Server
The STUN servers to use. See Reference12r1:IP4/General/STUN for details regarding the format.
TURN Server
The IP address or host name of the TURN server to use.
TURN Username
The user name for communication with the TURN server.
TURN Password
The password for communication with the TURN server.

General Coder Preference

This configuration applies to outgoing calls.

Coder
The protocols for voice compression are listed in this parameter.
Framesize [ms]
The packet sizes are automatically negotiated with 60 ms. With this parameter, you can specify a different RTP packet size.
Silence compression
No outbound packets are transmitted from the IP telephone during silence (no conversation).
Audio Only
Disables Video Codec negotiation. Do not affect calls with Local Network Coder. This option enables video calls only on Local Networks.
Exclusive
This parameter allows no negotiation, but accepts only the preset parameters.

Local Network Coder

This configuration applies to incoming calls and only if the remote RTP address is considered to be a local network address. See Reference:Configuration/IP/Settings for information on how to declare local network addresses.

Coder
The protocols for voice compression are listed in this parameter.
Framesize [ms]
The packet sizes are automatically negotiated with 60 ms. With this parameter, you can specify a different RTP packet size.
Silence compression
No outbound packets are transmitted from the IP telephone during silence (no conversation).

SRTP Cipher

This is the SRTP cipher suite that is offered for outgoing calls. If the remote party also provides an SRTP encryption key, media streaming will be encrypted. This setting applies only for SDES key exchange (see below).

SRTP Key Exchange

Two different key exchange methods for SRTP can be selected. SDES is hop-to-hop inside the signalling. DTLS-SRTP is inband and end-to-end. DTLS-SRTP is more secure but adds an additional delay to the beginning of phone calls. In the following "DTLS" means "DTLS-SRTP".

This setting has an influence on what key exchange methods are offered for outgoing calls and what method is selected on incoming calls.

SDES-DTLS
Both are offered. SDES is selected, if possible.
DTLS-SDES
Both are offered. DTLS is selected, if possible.
SDES
Only SDES is offered. SDES is selected, if possible.
DTLS
Only DTLS is offered. DTLS is selected, if possible.
No encryption
No SRTP is offered. No SRTP is selected.

No DTMF Detection

  • DTMF tones are sent in-band through the media channel but not as separate signalling messages (that is, they are not signalled at all, neither in-band nor out-band but sent as plain voice).

No Physical Location

  • The phone redirected from one PBX to another will not provide the physical location information to the redirected PBX, if this option is activated. This option disables the local flag functionality for specific endpoints (refer the Local option in Reference12r1:PBX/Objects).

Record to (URL)

HTTP URL where the recording file is to be stored. HTTP server must allow write access (PUT) at this location. One PCAP file is written for every call via this interface containing both RTP streams. Audio streams can be played using Wireshark. PCAP file can be converted into WAV file using pcap2wav tool. The filename of the recording file has the form <conference ID>-<mac>-<seq>.pcap

conference ID
A guid identifying the call across different hops. It is forwarded within the signaling. The CDRs generated by the PBX contain this conference ID.
mac
The Mac address of the device recording the call. Together with seq it is used to make sure the filename is unique if separate devices record the same call.
seq
A sequence number of the record starting with 0 after power up of the recording device. Together with mac it is used to make sure the filename is unique, especially to cover the case, that one device records the same call multiple times (e.g. incoming and outgoing)

Remark: Functionality of this field is obsolete for phones with low memory (such as IP230, IP240) running with firmware V11r2 or higher due to missing webDAV-service.

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