Web Interface/Advanced/Advanced Network
From Snom User Wiki
Dynamic RTP port start [rtp_port_start]
- Web User Interface:
- If you want to set up the port range out of which the RTP ports will be dynamically taken, specify the start port and end port number, respectively, in these fields.
- Valid values: Valid port numbers.
- Default value: blank
Link:
Dynamic RTP port stop [rtp_port_end]
- Web User Interface:
- Description: If you want to set up the port range out of which the RTP ports will be dynamically taken, specify the start port and end port number, respectively, in these fields.
- Valid values: Valid port numbers.
- Default value: blank
RTP Type of Service (TOS/Diffserv) [codec_tos]
- Web User Interface:
- Description: This option enables the phone to support quality of service (QOS) for RTP traffic in a network. This makes sense only if all parts of the involved network also support QOS.
- Valid values: Between <0> and <255>
- Default value: 160
SIP Type of Service (TOS/Diffserv) [signaling_tos]
- Web User Interface:
- Description: This option enables the phone to support quality of service (QOS) for SIP traffic in a network. This makes sense only if all parts of the involved network also support QOS.
- Valid values: Between <0> and <255>
- Default value: 160
DTMF Payload Type [dtmf_payload_type]
- Web User Interface:
- Description: Set up the payload type for Out-of-Band DTMF here The default setting is 101. This can be an arbitrary 8-bit value as long as the involved communication partners are both using the same value.
- Valid values: Integer values, e.g. <100>, <150>.
- Default value: 101
Network identity (port) [network_id_port]
- Web User Interface:
- Description: Set a static local port number, which is used for the SIP protocol communication, in this field. Usually, the phone chooses a random one!
- Valid values: Valid port number.
- Default value: blank
SIP T1 (ms) [sip_retry_t1]
- Web User Interface:
- Description: Set the retry timer in milliseconds after which an unanswered request is resent. If it is set to 500, the phone will resend the unanswered request after 500, 1000, 2000, 4000, 6000 ... 31500 ms. If the request is still unanswered after this procedure, an error message will be shown on the display.
- Valid values: Integer values, e.g. <500>.
- Default value: 500, 4000
Timer Support (RFC4028) [enable_timer_support]
- Web User Interface:
- Description: When set to <on> this setting enables the session timer according to [RFC4028].
- Valid values <on>, <off>
- Default value: on
SIP Session Timer (s) [session_timer]
- Web User Interface:
- Description: If SIP Session Timer Support is enabled, this option specifies the SIP session timer in seconds. For instance, a Re-INVITE will be sent after 50% of its value has elapsed.
- Valid values: Integer values, e.g. <0>, <2400>, <3200>.
- Default value: 3600 [sec]
- Link: Timer Support
SIP Dirty Host TTL (s) [dirty_host_ttl]
- Web User Interface:
- Description: Specify the “Time to Live” (TTL) for dirty hosts in seconds. This means that, when a phone was unable to reach a host, the phone will not try to reach this host again until the time specified in this field has elapsed.
- Valid values: Integer values, e.g. <60>, <120>.
- Default value: 0
SIP Max Forwards [max_forwards ]
- Web User Interface:
- Description: If you set a maximum number of forwards in this field, each time a forward is sent the counter is reduced by one. When zero is reached, the forwarding will stop. This prevents the phone from running into a SIP message-forwarding loop.
- Valid values: Integer values, e.g. <40>, <60>.
- Default value: 70
ENUM Suffix [enum_suffix]
- Web User Interface:
- Description: When using ENUM, you can specify a service suffix here, if desired. There is more than one service that supports ENUM lookups, and you can select here which one you want to use. Leave the default value e164.arpa if you don’t know better.
- Valid values: e.g. <e164.arpa>
- Default value: e164.arpa
- Note: Comma separated list of route domains for ENUM lookup, see also ENUM on snom phones.
Use user=phone [user_phone ]
- Web User Interface:
- Description: Turn this setting on if you want to use user=phone in SIP URIs. This is to distinguish phones from different non-phone devices like gateways, etc. (RFC 2543 deprecated).
- Valid values: <on>, <off>
- Default value: on
Publish Presence [publish_presence]
- Web User Interface:
- Description: When this feature is set to “on”, the phone sends out PUBLISH SIP messages showing the phone’s status.
- Valid values: <on>, <off>
- Default value: off
Refer-To Brackets [refer_brackets ]
- Web User Interface:
- Description: Switch additional brackets on or off in the Signaling for Refer-To. As some devices rely on this setting, we are kind enough to offer a solution!
- Valid values: <on>, <off>
- Default value: off
Require PRACK [require_prack]
- Web User Interface:
- Description: To force the use of PRACK, choose “on” here. “PRACK” messages are used to acknowledge the receipt of “180 Ringing” messages, which are usually not acknowledged. This helps, for example, to inform gateways whether the phone has actually begun to ring.
- Valid values: <on>, <off>
- Default value: on
Offer GRUU [offer_gruu]
- Web User Interface:
- Description: This setting is used to toggle the support for GRUU (Globally Routable User agent URLs) in SIP. When several phones have the same account, each one of them can be identified by the proxy through this GRUU ID, which is unique for each phone and stays the same even after reboot.
- Valid values: <on>, <off>
- Default value: on
Offer MPO [offer_mpo]
- Web User Interface:
- Description: Using this setting, the user can turn the Media Path Optimization support on or off. Turning it on makes sense only when you have MPO-supporting session border controller devices in your environment (e.g., Jasomi).
- Valid values: <on>, <off>
- Default value: off
Filter packets from Registrar [filter_registrar]
- Web User Interface:
- Description: If set to “on”, all SIP packets not coming from the registrar/proxy will be ignored. For security reasons, “on” is the default setting. This may cause big problems in an environment where SIP packets from other sources also have to be accepted for proper functionality!
- Valid values: <on>, <off>
- Default value: on
- Note: You have to disable it to make a call flow work which isn't going via the proxy only !
Authentication for SIP Reboot [challenge_reboot]
- Web User Interface:
- Description: This setting enables and disables challenge responses for remote reboot requests.
- Valid values: <on>, <off>
- Default value: off.
Authentication for SIP Check-Sync [challenge_checksync]
- Web User Interface:
- Description: Turning this setting on enables challenge responses for Check-Sync requests.
- Valid values: <on>, <off>
- Default value: off
Use SIP Compact Headers [short_form]
- Web User Interface:
- Description: In order to let the phone generate short compact SIP headers this option should be enabled. Otherwise the old usual style of SIP headers will be generated.
- Valid values: <on>, <off>
- Default value: off
- Web User Interface:
- Description: If enabled, the phone uses the Real Time Control Protocol (RTCP) to measure the quality of the audio (RTP) streams.
- Valid values: <on>, <off>
- Default value: on
Listen on SIP TCP port [tcp_listen]
- Web User Interface:
- Description: By default the phone isn't listening on port 5060 for TCP connections, to change this behavior, enable this option.
- Valid values: <on>, <off>
- Default value: off
Delay Subscriptions [subscription_delay]
- Web User Interface:
-
: from Version 6.5.1 Delay Subscriptions
-
: from Version 7.3.4 in the settings file only
-
- Description: Selects a random number around the given value in seconds to send delayed batch subscriptions. Useful at bootup for certain servers. Its not set by default.
- Valid values: Integer values, e.g. <60>, <80>.
- Default value: 0
Subscription Expiry [subscription_expiry]
- Firmware Version:
from Version 6.5.1
- Description: Defines the expiration time (in seconds) of an subscription. The value is used in a SIP SUBSCRIBE message the following way:
Sent to ... SUBSCRIBE sip:extension@registrar SIP/2.0 ... CSeq: 3 SUBSCRIBE ... Event: dialog Accept: application/dialog-info+xml Expires: 3600 Content-Length: 0
The registrar (SIP Server) may confirm with a distinct value.
Received from ... SIP/2.0 200 Ok ... CSeq: 3 SUBSCRIBE ... Expires: 180 Content-Length: 0
- Valid values: Integer values, e.g. <30>, <2400>, <3200>.
- Default value: 3600 [sec]
Categories: Setting | RTP | Quality Of Service | DTMF | SIP | Require Reboot | ENUM | Presence | Security | RTCP | Extension Monitoring | BLF
