Avaya B179 SIP Conference Phone
Installation and Administration Guide
51
<require_100rel>
Specify whether support for reliable provisional response (100rel and PRACK)
should be required by default. Note that this setting can be further customized
in account configuration.
<use_srtp>
Specify default value of secure media transport usage. Note that this setting
can be further customized in account configuration.
0: SRTP will be disabled, and the transport will reject RTP/SAVP offer.
1: SRTP will be advertised as optional and incoming SRTP offer will be accepted.
2: The transport will require that RTP/SAVP media shall be used.
<srtp_secure_signaling>
Specify whether SRTP requires secure signalling. This option is only used
when use_srtp option above is non-zero. Note that this setting can be further
customized in account configuration.
0: SRTP does not require secure signalling
1: SRTP requires secure transport such as TLS
2: SRTP requires secure end-to-end transport (SIPS)
<codec>
<type>
Codec type
<name>
Codec name
<prio>
Codec priority (0-4)
<dtmf>
DTMF signalling. Default is 2.
0: In-band
1: SIP message
2: RTP message
<no_vad>
Disable VAD. Default is VAD enabled.
<ec_tail>
Echo canceller tail length, in milliseconds.
<enable_ice>
Enable ICE?
<enable_relay>
Enable ICE relay?
<enable_presence>
Enable the use of presence signalling.
<enable_sip_replaces>
<enable_blind_transfer>
<allow_contact_rewrite>
<tls>
<tls_password />
Password for the private key
<tls_method>
TLS protocol method from pjsip_ssl_method, which can be:
0: Default (SSLv23)
1: TLSv1
2: SSLv2
3: SSLv3
23: SSLv23
<tls_verify_server>
Verify server certificate.
<tls_verify_client>
Verify client certificate.
<tls_require_client_cert>
Require client certificate.
<tls_neg_timeout>
TLS negotiation timeout in seconds to be applied for both outgoing and
incoming connections. If zero, no timeout is used.
<account1>
<valid>
If this account information is valid or not.