Web Interface/Advanced/Advanced Network

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Web User Interface: : Advanced - Advanced Network (Section): Dynamic RTP port start; : Advanced - SIP/RTP (Tab) - RTP/RTCP(Section):Dynamic RTP port start; : Advanced - SIP/RTP (Tab) - RTP/RTCP(Section):Dynamic RTP port start
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.
VALIDVALUE: Integer, valid port numbers
DEFAULTVALUE: 49152

Web User Interface: : Advanced - Advanced Network (Section): Dynamic RTP port stop; : Advanced - SIP/RTP (Tab) - RTP/RTCP(Section):Dynamic RTP port stop; : Advanced - SIP/RTP (Tab) - RTP/RTCP(Section):Dynamic RTP port stop
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.
VALIDVALUE: Integer, valid port numbers.
DEFAULTVALUE: 65534

Web User Interface: : Advanced - Advanced Network (Section): RTP Type of Service (TOS/Diffserv); : Advanced - QoS/Security (Tab) -Quality Of Service(Section):RTP Type of Service (TOS/Diffserv); : Advanced - QoS/Security (Tab) -Quality Of Service(Section):RTP Type of Service (TOS/Diffserv)
XML Syntax: Settings/Advanced Network/xml
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.
VALIDVALUE: Between <0> and <255>
DEFAULTVALUE: 160
FURTHER INFORMATION: FAQ: What does the TOS value 160 mean; Codecs

Web User Interface: : Advanced - Advanced Network (Section): SIP Type of Service (TOS/Diffserv)); : Advanced -QoS/Security (Tab) -Quality Of Service(Section):SIP Type of Service (TOS/Diffserv)); : Advanced -QoS/Security (Tab) -Quality Of Service(Section):SIP Type of Service (TOS/Diffserv))
XML Syntax: Settings/Advanced Network/xml
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.
VALIDVALUE: Between <0> and <255>
DEFAULTVALUE: 160

Web User Interface: : Advanced - Advanced Network (Section): DTMF Payload Type; : Advanced - SIP/RTP (Tab) - RTP/RTCP(Section):DTMF Payload Type; : Advanced - SIP/RTP (Tab) - RTP/RTCP(Section):DTMF Payload Type
XML Syntax: Settings/Advanced Network/xml
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.
VALIDVALUE: Integer values, e.g. <100>, <150>.
DEFAULTVALUE: 101

Web User Interface: : Advanced - Advanced Network (Section): Network identity (port); : Advanced - SIP/RTP (Tab) - SIP(Section):Network identity (port) (Reboot required); : Advanced - SIP/RTP (Tab) - SIP(Section):Network identity (port) (Reboot required)
XML Syntax: Settings/Advanced Network/xml
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!
VALIDVALUE: Integer (valid port number)
DEFAULTVALUE: blank
FURTHER INFORMATION: SIP

Web User Interface: : Advanced - Advanced Network (Section): SIP T1 (ms); : Advanced - SIP/RTP (Tab) - SIP(Section):SIP T1 (ms); : Advanced - SIP/RTP (Tab) - SIP(Section):SIP T1 (ms)
XML Syntax: Settings/Advanced Network/xml
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.
VALIDVALUE: Integer values, e.g. 500
DEFAULTVALUE: 500; 4000

Web User Interface: : Advanced - Advanced Network (Section): Timer Support (RFC 4028); : Advanced - SIP/RTP (Tab) - SIP(Section):Timer Support (RFC 4028); : Advanced - SIP/RTP (Tab) - SIP(Section):Timer Support (RFC 4028) ;XML Syntax
:Settings/Advanced Network/xml
Description: When set to <on> this setting enables the session timer according to RFC 4028.
VALIDVALUE: <on>, <off>
DEFAULTVALUE: on

Web User Interface: : Advanced - Advanced Network (Section): SIP Session Timer (s); : Advanced - SIP/RTP (Tab) - SIP(Section):SIP Session Timer (s); : Advanced - SIP/RTP (Tab) - SIP(Section):SIP Session Timer (s)
XML Syntax: Settings/Advanced Network/xml
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.
VALIDVALUE: Integer values, e.g. <0>, <2400>, <3200>.
DEFAULTVALUE: 3600 [sec]
Link: Timer Support

SIP Dirty Host TTL (s) [dirty_host_ttl]

Web User Interface: : Advanced - Advanced Network (Section): SIP Dirty Host TTL (s); : Advanced - SIP/RTP (Tab) - SIP(Section):SIP Dirty Host TTL (s); : Advanced - SIP/RTP (Tab) - SIP(Section):SIP Dirty Host TTL (s)
XML Syntax: Settings/Advanced Network/xml
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.
VALIDVALUE: Integer values, e.g. <60>, <120>.
DEFAULTVALUE: 0
FURTHER INFORMATION: TTL

SIP Max Forwards [max_forwards ]

Web User Interface: : Advanced - Advanced Network (Section): SIP Max Forwards; : Advanced - SIP/RTP (Tab) - SIP(Section):SIP Max Forwards; : Advanced - SIP/RTP (Tab) - SIP(Section):SIP Max Forwards
XML Syntax: Settings/Advanced Network/xml
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.
VALIDVALUE: Integer values, e.g. <40>, <60>.
DEFAULTVALUE: 70
FURTHER INFORMATION: SIP

Web User Interface: : Advanced - Advanced Network (Section): ENUM Sufﬁx; : Advanced - SIP/RTP (Tab) - SIP(Section):ENUM Sufﬁx; : Advanced - SIP/RTP (Tab) - SIP(Section):ENUM Sufﬁx
XML Syntax: Settings/Advanced Network/xml
Settings/Advanced Network/description
FURTHER INFORMATION: ENUM on snom phones

Web User Interface: : Advanced - Advanced Network (Section): Use user=phone; : Advanced - SIP/RTP (Tab) - SIP(Section):Use user=phone; : Advanced - SIP/RTP (Tab) - SIP(Section):Use user=phone
XML Syntax: Settings/Advanced Network/xml
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).
VALIDVALUE: <on>, <off>
DEFAULTVALUE: on

Web User Interface: : Advanced - Advanced Network (Section): Publish Presence; : Advanced - SIP/RTP (Tab) - SIP(Section):Publish Presence; : Advanced - SIP/RTP (Tab) - SIP(Section):Publish Presence
XML Syntax: Settings/Advanced Network/xml
Description: When this feature is set to “on”, the phone sends out PUBLISH SIP messages showing the phone’s status.
VALIDVALUE: <on>, <off>
DEFAULTVALUE: off

Web User Interface: : Advanced - Advanced Network (Section): Refer-To Brackets; : Advanced - SIP/RTP (Tab) - SIP(Section):Refer-To Brackets; : Advanced - SIP/RTP (Tab) - SIP(Section):Refer-To Brackets ;XML Syntax
:Settings/Advanced Network/xml
Description: Switch additional brackets on or off in the Signaling for Refer-To. Some devices rely on this setting.
VALIDVALUE: <on>, <off>
DEFAULTVALUE: off

Require PRACK [require_prack]

Web User Interface: : Advanced - Advanced Network (Section): Require PRACK; : Advanced - SIP/RTP (Tab) - SIP(Section):Require PRACK; : Advanced - SIP/RTP (Tab) - SIP(Section):Require PRACK ;XML Syntax
:Settings/Advanced Network/xml
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.
VALIDVALUE: <on>, <off>
DEFAULTVALUE: on

Web User Interface: : Advanced - Advanced Network (Section): Offer GRUU; : Advanced - SIP/RTP (Tab) - SIP(Section):Offer GRUU; : Advanced - SIP/RTP (Tab) - SIP(Section):Offer GRUU
XML Syntax: Settings/Advanced Network/xml
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.
VALIDVALUE: <on>, <off>
DEFAULTVALUE: on
FURTHER INFORMATION: GRUU

Offer MPO [offer_mpo]

Web User Interface: : Advanced - Advanced Network (Section): Offer MPO; : Advanced - SIP/RTP (Tab) - SIP(Section):Offer MPO; : Advanced - SIP/RTP (Tab) - SIP(Section):Offer MPO
XML Syntax: Settings/Advanced Network/xml
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).
VALIDVALUE: <on>, <off>
DEFAULTVALUE: off

Web User Interface: : Advanced - Advanced Network (Section): Filter packets from Registrar; : Advanced - QoS/Security (Tab) - Security(Section):Filter packets from Registrar; : Advanced - QoS/Security (Tab) - Security(Section):Filter packets from Registrar
XML Syntax: Settings/Advanced Network/xml
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! You have to disable it to make a call flow work which isn't going via the proxy only !
VALIDVALUE: on; off
DEFAULTVALUE: on
FURTHER INFORMATION: Security

Web User Interface: : Advanced - Advanced Network (Section): Authentication for SIP Reboot; : Advanced - QoS/Security (Tab) - Security(Section):Authentication for SIP Reboot; : Advanced - QoS/Security (Tab) - Security(Section):Authentication for SIP Reboot
Phone User Interface: Navigation Key Down - Configuration; Navigation Key right or press "Settings" - Configuration
XML Syntax: Settings/Advanced Network/xml
Description: This setting enables and disables challenge responses for remote reboot requests.
VALIDVALUE: <on>, <off>
DEFAULTVALUE: off.
FURTHER INFORMATION: Security

Web User Interface: : Advanced - Advanced Network (Section): Authentication for SIP Check-Sync; : Advanced - QoS/Security (Tab) - Security(Section):Authentication for SIP Check-Sync; : Advanced - QoS/Security (Tab) - Security(Section):Authentication for SIP Check-Sync
XML Syntax: Settings/Advanced Network/xml
Description: Turning this setting on enables challenge responses for Check-Sync requests.
VALIDVALUE: <on>, <off>
DEFAULTVALUE: off
FURTHER INFORMATION: Security

Web User Interface: : Advanced - Advanced Network (Section): Use SIP Compact Headers; : Advanced - SIP/RTP (Tab) - SIP(Section):Use SIP Compact Headers; : Advanced - SIP/RTP (Tab) - SIP(Section):Use SIP Compact Headers
XML Syntax: Settings/Advanced Network/xml
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.
VALIDVALUE: <on>, <off>
DEFAULTVALUE: off

Web User Interface: : Advanced - Advanced Network (Section): RTCP Support; : Advanced - SIP/RTP (Tab) - RTP/RTCP(Section): RTCP Support; : Advanced - SIP/RTP (Tab) - RTP/RTCP(Section): RTCP Support
Description: If enabled, the phone uses the Real Time Control Protocol (RTCP) to measure the quality of the audio (RTP) streams.
VALIDVALUE: <on>, <off>
DEFAULTVALUE: on

Web User Interface: : Advanced - Advanced Network (Section): Listen on SIP TCP port; : Advanced - SIP/RTP (Tab) - SIP(Section):Listen on SIP TCP port (Reboot required); : Advanced - SIP/RTP (Tab) - SIP(Section):Listen on SIP TCP port (Reboot required) ;XML Syntax
:Settings/Advanced Network/xml
Description: By default the phone isn't listening on port 5060 for TCP connections, to change this behavior, enable this option.
VALIDVALUE: <on>, <off>
DEFAULTVALUE: off

Web User Interface: : from Version 6.5.1 Delay Subscriptions; : from Version 7.3.4 via Auto Provisioning only; : via Auto Provisioning only
XML Syntax: Settings/Advanced Network/xml
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.
VALIDVALUE: Integer values, e.g. <60>, <80>.
DEFAULTVALUE: 0

Firmware Version: from Version 6.5.1
XML Syntax: Settings/Advanced Network/xml
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

VALIDVALUE: Integer values, e.g. <30>, <2400>, <3200>.
DEFAULTVALUE: 3600 [sec]

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Web Interface/Advanced/Advanced Network

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