User's Manual

Avaya B179 SIP Conference Phone Installation and Administration Guide
51
2: add both NAT type number and name
<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 conguration.
<use_srtp> Specify default value of secure media transport usage. Note that this setting
can be further customized in account conguration.
0: SRTP will be disabled, and the transport will reject RTP/SAVP oer.
1: SRTP will be advertised as optional and incoming SRTP oer 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 conguration.
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 certicate.
<tls_verify_client> Verify client certicate.
<tls_require_client_cert> Require client certicate.
<tls_neg_timeout> TLS negotiation timeout in seconds to be applied for both outgoing and
incoming connections. If zero, no timeout is used.