General SIP Parameters

Defines which IP address to use for classifying the incoming SIP dialog message to a Server-type IP Group, based on Proxy Set.

Note:

Defines the maximum number of characters allowed in the SDP body's "o=" (originator and session identifier) field for the session ID and session version values.

Below is an example of an "o=" line with session ID and session version values (in bold):

o=jdoe 2890844526 2890842807 IN IP4 10.47.16.5

The valid value range is 1,000 to 214,748,3647 (default).

Enables the device to unregister all user Accounts that were registered with the device, upon a device restart. During device start-up, each Account sends a REGISTER message (containing "Contact: *") to unregister all contact URIs belonging to its Address-of-Record (AOR), and then a second after they are unregistered, the device re-registers the Account.

To configure Accounts, see Configuring Registration Accounts.

Disables the sending of SIP 503 (Service Unavailable) responses upon receipt of new SIP dialog-initiating requests when the device's CPU is overloaded and thus, unable to accept and process new SIP messages.

Note: Even if the parameter is disabled (i.e., 503 is not sent), the device still discards the new SIP dialog-initiating requests when the CPU is overloaded.

Enables the device to send SIP 408 responses (Request Timeout) upon receipt of non-INVITE transactions. Disabling this response complies with RFC 4320/4321. By default, and in certain circumstances such as a timeout expiry, the device sends a SIP 408 Request Timeout in response to non-INVITE requests (e.g., REGISTER).

Enables a specific device action upon the receipt of a SIP NOTIFY request, where the action depends on the value in the Event header.

The action depends on the Event header value:

Note: The Event header value is proprietary to AudioCodes.

Defines the maximum size (in Kbytes) for each SIP message that can be sent over the network. The device rejects messages exceeding this user-defined size.

The valid value range is 1 to 100. The default is 100.

Determines whether the device sends SIP responses to the UDP port from where SIP requests are received even if the 'rport' parameter is not present in the SIP Via header.

Enables or disables the device to accept or reject SIP CANCEL requests received after the receipt of a 200 OK in response to an INVITE (i.e., call established). According to the SIP standard, a CANCEL can be sent only during the INVITE transaction (before 200 OK), and once a 200 OK response is received the call can be rejected only by a BYE request.

Defines how the device handles SIP Call-Info headers with multiple values in outgoing SIP messages.

Enables the device to reject SIP requests (e.g., ACK, BYE, or re-INVITE) whose user part in the Request-URI is different from the user part in the Contact header of the last sent SIP request.

The [VerifyRecievedRequestUri] parameter functions together with the [RegistrarProxySetID] parameter, as follows:

Note: This handling is applicable only to the SBC application.

Note: This handling is applicable to the Gateway and SBC applications.

Defines a Proxy Set for the registrar. The parameter functions together with the [VerifyRecievedRequestUri] parameter. For more information, see the description of the [VerifyRecievedRequestUri] parameter.

The default value is -1 (not defined).

Note: This setting assumes that the SIP Interface has only one registrar.

Defines the maximum number of simultaneous active calls supported by the device. If the maximum number of calls is reached, new calls are not established.

The valid range is 1 to the maximum number of supported channels. The default value is the maximum available channels (i.e., no restriction on the maximum number of calls).

Enables the device to include call Quality of Service (QoS) statistics in SIP BYE messages and SIP 200 OK responses to BYE messages, using the proprietary SIP header X-RTP-Stat.

The X-RTP-Stat header contains the following statistics:

The X-RTP-Stat header contains the following fields:

Below is an example of the X-RTP-Stat header in a SIP BYE message:

BYE sip:302@10.33.4.125 SIP/2.0

Via: SIP/2.0/UDP 10.33.4.126;branch=z9hG4bKac2127550866

Max-Forwards: 70

From: <sip:401@10.33.4.126;user=phone>;tag=1c2113553324

To: <sip:302@company.com>;tag=1c991751121

Call-ID: 991750671245200001912@10.33.4.125

CSeq: 1 BYE

X-RTP-Stat: PS=207;OS=49680;;PR=314;OR=50240;PL=0;JI=600;LA=40;

Supported: em,timer,replaces,path,resource-priority

Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE

User-Agent: Sip-Gateway-/7.40A.600.203

Reason: Q.850 ;cause=16 ;text="local"

Content-Length: 0

Note: The parameter is applicable only to the Gateway application.

Determines the PRACK (Provisional Acknowledgment) mechanism mode for SIP 1xx reliable responses.

Note:

Global parameter enabling the Early Media feature for sending media (e.g., ringing) before the call is established.

You can also configure this feature per specific calls, using IP Profiles ('Early Media' parameter) or Tel Profiles ('Enable Early Media' parameter). For a detailed description of the parameter and for configuring the feature, see Configuring IP Profiles or Configuring Tel Profiles.

Note: If the feature is configured for a specific profile, the settings of the global parameter is ignored for calls associated with the profile.

Global parameter that enables the device to send SIP 183 responses with SDP to the IP upon receipt of INVITE messages. You can also configure this feature per specific calls, using IP Profiles ('Early 183' parameter). For a detailed description of the parameter and for configuring this feature in the IP Profiles table, see Configuring IP Profiles.

Note: If you configure this feature for a specific IP Profile, the device ignores this global parameter for calls associated with the IP Profile.

Defines the device's interworking of Alerting messages for IP-to-Tel calls (ISDN). It determines whether the device sends a 180 Ringing response to the caller after the device sends a 183 Session Progress response to the caller. The 180 Ringing response indicates that the INVITE has been received by the ISDN side and that alerting is taking place (i.e., ISDN Progress message), indicating to the IP PBX to play a ringback tone. The 183 Session Progress response allows an early media session to be established prior to the call being answered, for example, to hear a ring tone, busy tone or recorded announcement.

Note:

Defines the ISDN message that is sent when the 183 Session Progress message is received for IP-to-Tel calls.

Note: The parameter is applicable only to the Gateway application.

Typically, if an Q.931 Progress message with a Cause is received from the PSTN for an outgoing IP-to-ISDN call and the [EnableEarlyMedia] parameter is set to 1 (i.e., the Early Media feature is enabled), the device interworks the Progress to 183 + SDP to enable the originating party to hear the PSTN announcement about the call failure. Conversely, if EnableEarlyMedia is set to 0, the device disconnects the call by sending a SIP 4xx response to the originating party. However, if the [ReleaseIP2ISDNCallOnProgressWithCause] parameter is set to 1, then the device sends a SIP 4xx response even if the [EnableEarlyMedia] parameter is set to 1.

Note: The parameter is applicable only to digital interfaces.

Defines the numerical value sent in the Session-Expires header in the first SIP INVITE request or response (if the call is answered).

The valid range is 1 to 86,400 sec. The default is 0 (i.e., the Session-Expires header is disabled).

Note: The parameter is applicable only to the Gateway application.

Defines the time (in seconds) in the SIP Min-SE header. The header defines the minimum time that the user agent refreshes the session.

The valid range is 10 to 100,000. The default is 90.

Note: The parameter is applicable only to the Gateway application.

Defines a session expiry timeout.

The new session expiry timeout is calculated by subtracting the configured value from the original timeout as specified in the Session-Expires header. However, the new timeout must be greater than or equal to one-third (1/3) of the Session-Expires value. If the refresher doesn't send a refresh request within the new timeout, the device disconnects the session (i.e., sends a SIP BYE).

For example, if you configure the parameter to 32 seconds and the Session-Expires value is 180 seconds, the session timeout occurs 148 seconds (i.e., 180 minus 32) after the last session refresh. If the Session-Expires header value is 90 seconds, the timeout occurs 60 seconds after the last refresh. This is because 90 minus 32 is 58 seconds, which is less than one third of the Session-Expires value (i.e., 60/3 is 30, and 90 minus 30 is 60).

The valid range is 0 to 32 (in seconds). The default is 32.

Note: The parameter is applicable only to the Gateway application.

Defines the SIP method used for session-timer updates.

Note:

Determines whether the device removes the ‘to’ header tag from final SIP failure responses to INVITE transactions.

Enables the use of the 'rtcp' attribute in the outgoing SDP.

Note: The parameter is applicable only to the Gateway application.

Defines the user part value of the Request-URI for outgoing SIP OPTIONS requests. If no value is configured, configuration parameter ‘Username’ value is used.

A special value is ‘empty’, indicating that no user part in the Request-URI (host part only) is used.

The valid range is a 30-character string. By default, this value is not defined.

Enables the device to not respond to received SIP OPTIONS messages from, and/or not to send keep-alive messages to, a proxy server associated with Trunk Group ID 1 if all its member trunks are down.

Note:

Defines the minimal number of SIP dialogs that must be established when using TDM Tunneling, for the specific trunk to be considered active.

When using TDM Tunneling, if calls from this defined number of B-channels pertaining to a specific Trunk fail (i.e., SIP dialogs are not correctly set up), an AIS alarm is sent on this trunk toward the PSTN and all current calls are dropped. The originator gateway continues the INVITE attempts. When this number of calls succeed (i.e., SIP dialogs are correctly set up), the AIS alarm is cleared.

The valid range is 0 to 31. The default is 0 (i.e., don't send AIS alarms).

Note: TDM Tunneling is applicable only to E1/T1 interfaces.

Defines the time (in msec) between the first INVITE issued within the same trunk when implementing the TDM tunneling application.

The valid value range is 500 to 1,000. The default is 500.

Note: TDM Tunneling is applicable only to E1/T1 interfaces.

Defines the time (in msec) between call release and a new INVITE when implementing the TDM tunneling application.

The valid value range is 10,000 to 20,000. The default is 10,000.

Note: TDM Tunneling is applicable only to E1/T1 interfaces.

Global parameter defining the SIP signaling method for establishing and transmitting a fax session when the device detects a fax.

You can also configure this feature per specific calls, using IP Profiles ('Fax Signaling Method' parameter) and Tel Profiles ('Fax Signaling Method' parameter). For a detailed description of the parameter, see Configuring IP Profiles and Configuring Tel Profiles.

Note: If you configure this feature for a specific IP Profile or Tel Profile, the device ignores this global parameter for calls associated with the IP Profile or Tel Profile.

Determines the device's fax transport behavior when G.711 VBD coder is negotiated at call start.

Note:

Defines the payload type of the outgoing SDP offer.

The valid value range is 96 to 127 (dynamic payload type). The default is 0 (i.e. NoAudio is not supported). For example, if set to 120, the following is added to the INVITE SDP:

a=rtpmap:120 NoAudio/8000\r\n

Note: For incoming SDP offers, NoAudio is always supported.

Determines the default transport layer for outgoing SIP calls initiated by the device.

Note:

Enables the device to add the default SIP port 5060 (UDP/TCP) or 5061 (TLS) to outgoing messages that are received without a port. This condition also applies to manipulated messages where the resulting message has no port number. The device adds the default port number to the following SIP headers: Request-Uri, To, From, P-Asserted-Identity, P-Preferred-Identity, and P-Called-Party-ID. If the message is received with a port number other than the default, for example, 5070, the port number is not changed.

An example of a SIP From header with the default port is shown below:

From: <sip:+4000@10.8.4.105:5060;user=phone>;tag=f25419a96a;epid=009FAB8F3E

Enables secured SIP (SIPS URI) connections over multiple hops.

When the [SIPTransportType] parameter is set to 2 (i.e., TLS) and the parameter [EnableSIPS] is disabled, TLS is used for the next network hop only. When the [SIPTransportType] parameter is set to 2 or 1 (i.e., TCP or TLS) and [EnableSIPS] is enabled, TLS is used through the entire connection (over multiple hops).

Note: If the parameter is enabled and the [SIPTransportType] parameter is set to 0 (i.e., UDP), the connection fails.

Enables the reuse of an established TCP or TLS connection between the device and a SIP user agent (UA) for subsequent SIP requests sent to the UA. Any new out-of-dialog requests (e.g., INVITE or REGISTER) use the same secured connection. One of the benefits of enabling the parameter is that it may improve performance by eliminating the need for additional TCP/TLS handshakes with the UA, allowing sessions to be established rapidly.

Note:

Enables the reuse of the same TCP/TLS connection for sessions with the same user even if the 'alias' parameter is not present in the SIP Via header of the initial INVITE.

Note: To enable TCP/TLS connection reuse, see the [EnableTCPConnectionReuse] parameter.

Enables all reusable TCP/TLS (reliable) connections to be persistent (i.e., not released).

When sending a SIP message, the device’s reliable connection reuse policy determines if current connections to the specific destination are reused.

Persistent connections ensure less network traffic due to fewer setting up and tearing down of reliable connections and reduced latency on subsequent requests because there is no need for initial TCP handshakes. Persistent connections may reduce the number of costly TLS handshakes to establish security associations, in addition to the initial TCP connection setup.

Note:

Defines the Timer B (INVITE transaction timeout timer) and Timer F (non-INVITE transaction timeout timer), as defined in RFC 3261, when the SIP transport type is TCP.

The valid range is 0 to 60 sec. The default is 0, which means that the parameter's value is set to 64 multiplied by the value of the [SipT1Rtx] parameter. For example, if you configure [SipT1Rtx] to 500 msec (0.5 sec) and leave the [SIPTCPTimeout] parameter at its default value (0), the actual value of [SIPTCPTimeout] is 32 sec (64 x 0.5 sec).

Defines the SIP destination port for sending initial SIP requests.

The valid range is 1 to 65534. The default port is 5060.

Note: SIP responses are sent to the port specified in the Via header.

Defines if the 'user=phone' string is added to the SIP URI and SIP To header.

Note: The parameter is applicable only to the Gateway application.

Defines if the 'user=phone' string is added to the From and Contact SIP headers.

Note: The parameter is applicable only to the Gateway application.

Defines the format of the URI in the P-Asserted-Identity and P-Preferred-Identity headers.

Defines the number of P-Preferred-Identity SIP headers included in the outgoing SIP message when the header contains multiple values.

P-Preferred-Identity: <sip:someone@test.org>,<tel:+123456789>

P-Preferred-Identity: <sip:someone@test.org>
P-Preferred-Identity: <tel:+123456789> 

P-Preferred-Identity: <sip:someone@test.org>
P-Preferred-Identity: <tel:+123456789> 

P-Preferred-Identity: <sip:someone@test.org>,<tel:+123456789>

Note:

Defines the time (in seconds) that the device waits for a 200 OK response from the called party (IP side) after sending an INVITE message, for Tel-to-IP calls. If the timer expires, the call is released.

The valid range is 0 to 3600. The default is 180.

Enables Remote-Party-Identity headers for calling and called numbers for Tel-to-IP calls.

Enables usage of the SIP History-Info header.

User Agent Client (UAC) Behavior:

User Agent Server (UAS) Behavior:

Note: The parameter is applicable only to digital interfaces.

Enables the device to add the SIP 'tgrp' parameter to outgoing SIP message requests. This parameter specifies the Trunk Group to which the call belongs (according to RFC 4904). For example, the INVITE message below indicates that the call belongs to Trunk Group ID 1:

INVITE sip::+16305550100;tgrp=1;trunk-context=example.com@10.1.0.3;user=phone SIP/2.0

INVITE sip:1234567;tgrp=hotline-ccdata;trunk-context=dsn.mil@example.com

Note: IP-to-Tel configuration (see Configuring IP-to-Tel Routing Rules) overrides the 'tgrp' parameter in incoming INVITE messages.

Defines the precedence method for IP-to-Tel call routing - according to the IP-to-Tel Routing table or the SIP 'tgrp' parameter.

The following example shows an INVITE Request-URI with the 'tgrp' parameter, indicating that the IP call should be routed to Trunk Group 7:

INVITE sip:200;tgrp=7;trunk-context=example.com@10.33.2.68;user=phone SIP/2.0

Note:

Defines if the device uses the SIP 'dtg' parameter for routing IP-to-Tel calls to a specific Trunk Group.

When enabled, if the Request-URI in the received SIP INVITE includes the 'dtg' parameter, the device routes the call to the Trunk Group according to its value. The parameter is used instead of the 'tgrp' and 'trunk-context' parameters. The 'dtg' parameter appears in the INVITE Request-URI (and in the To header). For example, the following received SIP message routes the call to Trunk Group ID 56:

INVITE sip:123456@192.168.1.2;dtg=56;user=phone SIP/2.0

Note: If the Trunk Group is not found based on the 'dtg' parameter, the device uses the IP-to-Tel Routing table to route the call to the appropriate Trunk Group.

Determines whether the Globally Routable User Agent URIs (GRUU) mechanism is used, according to RFC 5627. This is used for obtaining a GRUU from a registrar and for communicating a GRUU to a peer within a dialog.

A GRUU is a SIP URI that routes to an instance-specific UA and can be reachable from anywhere. There are a number of contexts in which it is desirable to have an identifier that addresses a single UA (using GRUU) rather than the group of UA’s indicated by an Address of Record (AOR). For example, in call transfer where user A is talking to user B, and user A wants to transfer the call to user C. User A sends a REFER to user C:

REFER sip:C@domain.com SIP/2.0

From: sip:A@domain.com;tag=99asd

To: sip:C@domain.com

Refer-To: (URI that identifies B's UA)

The Refer-To header needs to contain a URI that user C can use to place a call to user B. This call needs to route to the specific UA instance that user B is using to talk to user A. User B should provide user A with a URI that has to be usable by anyone. It needs to be a GRUU.

If the remote server doesn’t support GRUU, it ignores the parameters of the GRUU. Otherwise, if the remote side also supports GRUU, the REGISTER responses contain the “gruu” parameter in each Contact header. The parameter contains a SIP or SIPS URI that represents a GRUU corresponding to the UA instance that registered the contact. The server provides the same GRUU for the same AOR and instance-id when sending REGISTER again after registration expiration. RFC 5627 specifies that the remote target is a GRUU target if its’ Contact URL has the "gr" parameter with or without a value.

Determines whether a Cisco gateway exists at the remote side.

When a Cisco gateway exists at the remote side, the device must set the value of the 'annexb' parameter of the fmtp attribute in the SDP to 'no'. This logic is used if the coder is enabled for Silenced Suppression. In this case, Silence Suppression is used on the channel but not declared in the SDP.

Note:

Defines the string that is used in the SIP User-Agent and Server response headers. When configured, the string <UserAgentDisplayInfo value>/software version' is used, for example:

User-Agent: myproduct/7.40A.600.203

If not configured, the default string, "<product-name>/<<software version>>" is used, for example:

User-Agent: AudioCodes-Sip-Gateway/<swver>

The maximum string length is 50 characters.

Note: The software version number and preceding forward slash (/) cannot be modified. Therefore, it is recommended not to include a forward slash in the parameter's value (to avoid two forward slashes in the SIP header, which may cause problems).

Defines the value of the Owner line ('o' field) in outgoing SDP messages.

The valid range is a string of up to 39 characters. The default is "AudioCodesGW".

For example:

o=AudioCodesGW 1145023829 1145023705 IN IP4 10.33.4.126

Note: The parameter is applicable only to the SBC application when the device creates a new SIP message (and SDP) such as when the device plays a ringback tone. The parameter is not applicable to SIP messages that the device receives from one end and sends to another (i.e., doesn't modify the SDP's 'o' field).

Enables the device to ignore new SDP re-offers (from the media negotiation perspective) in certain scenarios (such as session expires). According to RFC 3264, once an SDP session is established, a new SDP offer is considered a new offer only when the SDP origin value is incremented. In scenarios such as session expires, SDP negotiation is irrelevant and thus, the origin field is not changed.

Even though some SIP devices don’t follow this behavior and don’t increment the origin value even in scenarios where they want to re-negotiate, the device can assume that the remote party operates according to RFC 3264, and in cases where the origin field is not incremented, the device doesn't re-negotiate SDP capabilities.

Defines how the device displays the Connection ("c=") line in the SDP Offer/Answer model.

Note: The parameter is applicable only to the Gateway application.

Defines the Subject header value in outgoing INVITE messages. If not specified, the Subject header isn't included (default).

The maximum length is up to 50 characters.

Defines the priority for coder negotiation in the incoming SDP offer, between the device's or remote UA's coder list.

Note: The parameter is applicable only to the Gateway application.

Enables coder re-negotiation in the sent re-INVITE for retrieving an on-hold call.

The parameter is useful in the following call scenario example:

Note: The parameter is applicable only to the Gateway application.

Determines whether the 'mptime' attribute is included in the outgoing SDP.

The mptime' attribute enables the device to define a separate packetization period for each negotiated coder in the SDP. The 'mptime' attribute is only included if the parameter is enabled even if the remote side includes it in the SDP offer. Upon receipt, each coder receives its 'ptime' value in the following precedence: from 'mptime' attribute, from 'ptime' attribute, and then from default value.

Note: The parameter is applicable only to the Gateway application.

Defines if the 'ptime' attribute is included in the SDP.

Determines the device's behavior regarding call identifiers when a 3xx response is received for an outgoing INVITE request. The device can use the same call identifiers (Call-ID, To, and From tags) or change them in the new initiated INVITE.

Enables the inclusion of the P-Charging-Vector header to all outgoing INVITE messages.

Note: The parameter is applicable only to the Gateway application.

Defines the device’s behavior when it receives a SIP 503 (Service Unavailable) containing a Retry-After header, in response to a SIP message (e.g., REGISTER) sent to a proxy server.

In certain scenarios (depending on the value of this parameter), the device considers the proxy as offline (down) for the number of seconds specified in the Retry-After header. During this timeout, the device doesn't send any SIP messages to the proxy. This condition is indicated in the syslog message as "server is now Unavailable - setting Retry-After timer to x secs".

Defines the time (in seconds) used in the Retry-After header when a 503 (Service Unavailable) response is generated by the device.

The time range is 0 to 3,600. The default is 0.

Defines if the device, upon receiving a SIP 503 response without a Retry-After header, behaves as if the 503 response included a Retry-After header and with the period (in seconds) specified by the parameter.

When enabled, this feature allows the device to operate with Proxy servers that do not include the Retry-After SIP header in SIP 503 (Service Unavailable) responses to indicate an unavailable service.

The Retry-After header is used with the 503 (Service Unavailable) response to indicate how long the service is expected to be unavailable to the requesting SIP client. The device maintains a list of available proxies, by using the Keep-Alive mechanism. The device checks the availability of proxies by sending SIP OPTIONS every keep-alive timeout to all proxies.

If the device receives a SIP 503 response to an INVITE, it also marks that the proxy is out of service for the defined "Retry-After" period.

Determines the device usage of the P-Associated-URI header. This header can be received in 200 OK responses to REGISTER requests. When enabled, the first URI in the P-Associated-URI header is used in subsequent requests as the From/P-Asserted-Identity headers value.

Note: P-Associated-URIs in registration responses is handled only if the device is registered per endpoint (using the User Information file).

Defines if the destination phone number that the device sends to the Tel side (for IP-to-Tel calls), includes the user-part parameters (e.g., 'password' and 'phone-context') of the destination URI received in the incoming SIP INVITE message.

Note: The parameter is applicable only to the Gateway application.

Defines the SIP header from which the source (calling) number is obtained in incoming INVITE messages.

Note:

Enables the device to include all previously negotiated media lines ('m=') within the current session in the SDP offer-answer exchange (RFC 3264).

For example, assume a call (audio) has been established between two endpoints and one endpoint wants to subsequently send an image in the same call session. If the parameter is enabled, the endpoint includes the previously negotiated media type (i.e., audio) with the new negotiated media type (i.e., image) in its SDP offer:

v=0
o=bob 2890844730 2890844731 IN IP4 host.example.com
s=
c=IN IP4 host.example.com
t=0 0
m=audio 0 RTP/AVP 0
m=image 12345 udptl t38

In this example, if the parameter is disabled, the only ‘m=’ line included in the SDP is the newly negotiated media (i.e., image).

Note: The parameter is applicable only to the Gateway application.

Defines if the device sends a second source (calling) number, obtained from the incoming SIP INVITE message, to the Tel side.

The valid value is “P-Asserted” (without quotation marks). By default, no value is defined.

If the parameter is not configured to any value (i.e., default) or configured to any value other than "P-Asserted", the device doesn’t send a second source number. If the parameter is configured to "P-Asserted" and the incoming INVITE message contains a P-Asserted-Identity header(s), the device sends a second source number that is obtained from the first listed P-Asserted-Identity header in the message. If the message doesn’t include a P-Asserted-Identity header, the device sends a second source number that it obtains from the first source number (i.e., same number).

Note: The parameter is applicable only to the Gateway application.

Defines the SIP header from which the called (destination) number is obtained for IP-to-Tel calls.

Note: The parameter is applicable only to the Gateway application.

Enables the usage of the SIP Reason header.

Defines a name for the device (e.g., device123.com), which is used as the host part for the SIP URI in the From header for outgoing messages. If not configured, the device's IP address is used instead (default).

The valid value is a string of up to 100 characters. By default, no value is defined.

Note:

Determines the device's response to an incoming SDP that includes an IP address of 0.0.0.0 in the SDP's Connection Information field (i.e., "c=IN IP4 0.0.0.0").

Determines whether the device sends the initial INVITE message with or without an SDP. Sending the first INVITE without SDP is typically done by clients for obtaining the far-end's full list of capabilities before sending their own offer. (An alternative method for obtaining the list of supported capabilities is by using SIP OPTIONS, which is not supported by every SIP agent.)

Defines if the device includes or excludes URI parameters for the Digest URI in the SIP Proxy-Authorization or Authorization headers of the request that the device sends in reply to a received SIP 401 (Unauthorized) or 407 (Proxy Authentication Required) response.

Below shows an example of a request with an Authorization header containing a Digest URI (shown in bold):

Authorization: Digest username="alice at AudioCodes.com",realm="AudioCodes.com",nonce="",response="",uri="sip:AudioCodes.com"

Enables the device to send "a=crypto" lines without the lifetime parameter in the SDP. For example, if the SDP contains "a=crypto:12 AES_CM_128_HMAC_SHA1_80 inline:hhQe10yZRcRcpIFPkH5xYY9R1de37ogh9G1MpvNp|2^31", it removes the lifetime parameter "2^31".

Defines the AES-256 encryption key for encrypting (and decrypting) the SIP header value.

The valid value is a string of 32 characters. By default, no value is defined.

For more information, see Configuring SIP Header Value Encryption.

Determines whether the device sets the Contact header of outgoing INVITE requests to ‘anonymous’ for restricted calls.

Defines if the device's IP address is used as the URI host part instead of "anonymous.invalid" in the INVITE's From header for Tel-to-IP calls.

The parameter may be useful, for example, for service providers who identify their SIP Trunking customers by their source phone number or IP address, reflected in the From header of the SIP INVITE. Therefore, even customers blocking their Caller ID can be identified by the service provider. Typically, if the device receives a call with blocked Caller ID from the PSTN side (e.g., Trunk connected to a PBX), it sends an INVITE to the IP with a From header as follows: "From: “anonymous” <anonymous@anonymous.invalid>". This is in accordance with RFC 3325. However, when the parameter is set to 1, the device replaces the "anonymous.invalid" with its IP address.

Defines a 'representative number' (up to 50 characters) that is used as the user part of the Request-URI in the P-Asserted-Identity header of an outgoing INVITE for Tel-to-IP calls.

The default is null.

Defines the source for the SIP URI set in the Refer-To header of outgoing REFER messages.

Enables the usage of the User Information, which is loaded to the device in the User Information Auxiliary file. For more information on User Information, see User Information File.

Note: For the parameter to take effect, a device restart is required.

Determines whether the device uses the value of the incoming SIP Reason header for Release Reason mapping.

Determines the device's behavior upon receipt of SIP Re-INVITE messages that include the SDP's 'silencesupp:off' attribute.

Note: The parameter is applicable only if the G.711 coder is used.

Enables the usage of the 'rport' parameter in the Via header.

The device adds an 'rport' parameter to the Via header of each outgoing SIP message. The first Proxy that receives this message sets the 'rport' value of the response to the actual port from where the request was received. This method is used, for example, to enable the device to identify its port mapping outside a NAT.

If the Via header doesn't include the 'rport' parameter, the destination port of the response is obtained from the host part of the Via header.
If the Via header includes the 'rport' parameter without a port value, the destination port of the response is the source port of the incoming request.
If the Via header includes 'rport' with a port value (e.g., rport=1001), the destination port of the response is the port indicated in the 'rport' parmeter.

Enables the device to add the SIP X-Channel header to outgoing SIP messages. The header provides information on the physical Trunk/B-channel on which the call is received or sent.

x-channel: ds/ds1-<digital Trunk number>/<Trunk/B-channel>;IP=<device's IP address>

For example, the below shows a call on Trunk 1, channel 4 of the device with IP address 192.168.13.1:

x-channel: ds/ds1-1/4;IP=192.168.13.1

Global parameter defining the progress indicator (PI) sent to the IP.

You can also configure the feature per specific calls, using IP Profiles ('Progress Indicator to IP' parameter) or Tel Profiles ('Progress Indicator to IP' parameter). For a detailed description of the parameter and for configuring the feature, see Configuring IP Profiles or Configuring Tel Profiles.

Note: If you configure this feature for a specific profile, the device ignores this global parameter for calls associated with the profile.

Enables the device to send a re-INVITE with a new (different) SRTP key (in the SDP) if a SIP 181 response is received ("call is being forwarded"). The re-INVITE is sent immediately upon receipt of the 200 OK (when the call is answered).

Note: The parameter is applicable only if SRTP is used.

Defines the maximum number of concurrent, outgoing SIP REGISTER dialogs. The parameter is used to control the registration rate.

The valid range is 1 to 20. The default is 20.

Note:

Defines the Network Node Identifier of the device for Avaya UCID.

The valid value range is1 to 0x7FFF. The default is 0.

Note:

Defines the default Release Cause (sent to IP) for IP-to-Tel calls when the device initiates a call release and an explicit matching cause for this release is not found.

The default release cause is NO_ROUTE_TO_DESTINATION (3). Other common values include NO_CIRCUIT_AVAILABLE (34), DESTINATION_OUT_OF_ORDER (27), etc.

Note:

Enables the modification of the called and calling number for numbers received with Microsoft's proprietary "ext=xxx" parameter in the SIP INVITE URI user part. Microsoft Office Communications Server sometimes uses this proprietary parameter to indicate the extension number of the called or calling party.

For example, if a calling party makes a call to telephone number 622125519100 Ext. 104, the device receives the SIP INVITE (from Microsoft's application) with the URI user part as INVITE sip:622125519100;ext=104@10.1.1.10 (or INVITE tel:622125519100;ext=104). If the parameter [EnableMicrosofExt] is enabled, the device modifies the called number by adding an "e" as the prefix, removing the "ext=" parameter, and adding the extension number as the suffix (e.g., e622125519100104). Once modified, the device can then manipulate the number further, using the Number Manipulation tables to leave only the last 3 digits (for example) for sending to a PBX.

Defines the URI format in the SIP Diversion header.

Defines the timeout (in msec) between receiving a 100 Trying response and a subsequent 18x response. If a 18x response is not received within this timeout period, the call is disconnected.

The valid range is 0 to 180,000 (i.e., 3 minutes). The default is 32000 (i.e., 32 sec).

Determines whether the device ignores the Master Key Identifier (MKI) if present in the SDP received from the remote side.

Enables negotiation and usage of Comfort Noise (CN) for Gateway calls.

The use of CN is indicated by including a payload type for CN on the media description line of the SDP. The device can use CN with a codec whose RTP time stamp clock rate is 8,000 Hz (G.711/G.726). The static payload type 13 is used. The use of CN is negotiated between sides. Therefore, if the remote side doesn't support CN, it is not used. Regardless of the device's settings, it always attempts to adapt to the remote SIP UA's request for CNG, as described below.

To determine CNG support, the device uses the [ComfortNoiseNegotiation] parameter and the codec’s SCE (silence suppression setting) using the [CodersGroup] parameter.

If the [ComfortNoiseNegotiation] parameter is enabled, then the following occurs:

If the [ComfortNoiseNegotiation] parameter is disabled, then the device doesn't send “CN” in the SDP. However, if the codec’s SCE is enabled, then CNG occurs.

Note: The parameter is applicable only to the Gateway application.

Defines the echo canceller format in the outgoing SDP. The 'ecan' attribute is used in the SDP to indicate the use of echo cancellation.

Note: The parameter is applicable only when the [IsFaxUsed] parameter is set to 2, and for re-INVITE messages generated by the device as result of modem or fax tone detection.

Defines the index of the first ringback tone in the CPT file. This option enables an Application server to request the device to play a distinctive ringback tone to the calling party according to the destination of the call. The tone is played according to the Alert-Info header received in the 180 Ringing SIP response (the value of the Alert-Info header is added to the value of the parameter).

The valid range is -1 to 1,000. The default is -1 (i.e., play standard ringback tone).

Note:

Defines the time period the device waits for an MFC R2 Resume (Reanswer) signal once a Suspend (Clear back) signal is received from the PBX. If this timer expires, the call is released.

The valid range is 0 to 255 (in seconds). The default is 0.

Note: For digital interfaces, the parameter is applicable only to the MFC-R2 CAS Brazil variant.

The parameter is used for private wire services (see Configuring Private Wire Interworking).

Note:

Assigns the IP Group (by ID) configured for the Skype for Business Server (presence server). This is where the device sends SIP PUBLISH messages to notify of changes in presence status of Skype for Business users when making and receiving calls using third-party endpoint devices.

For more information on integration with Microsoft presence, see Microsoft Skype for Business Presence of Third-Party Endpoints.

Enables the device to notify (using SIP PUBLISH messages) Skype for Business Server (presence server) of changes in presence status of Skype for Business users when making and receiving calls using third-party endpoint devices.

For more information on integration with Microsoft presence, see Microsoft Skype for Business Presence of Third-Party Endpoints.

Defines the Alert Timeout (in seconds) for calls sent to the PSTN. This timer is used between the time a Setup message is sent to the Tel side (IP-to-Tel call establishment) and a Connect message is received. If an Alerting message is received, the timer is restarted. If the timer expires before the call is answered, the device disconnects the call and sends a SIP 408 request timeout response to the SIP party that initiated the call.

The valid value range is 1 to 600 (in seconds). The default is 180.

Note:If per trunk configuration, using the [TrunkPSTNAlertTimeout] parameter, is set to other than default, the [PSTNAlertTimeout] parameter value is overridden.

Enables the device to send and receive RTP packets to and from remote endpoints without the need to establish a SIP session. The remote IP address is determined according to the Tel-to-IP Routing table (Prefix parameter). The port is the same port as the local RTP port (configured by the [BaseUDPPort] parameter and the channel on which the call is received).

Note:

Enables the RTP Only feature per trunk. The x in the parameter name denotes the trunk number, where 0 is Trunk 1. For a description of the parameter, see the [RTPOnlyMode] parameter.

Note: For using the global parameter (i.e., setting the RTP Only feature for all trunks), set the parameter to -1 (default).

Global parameter that defines the preferred RTP media IP addressing version (IPv4 or IPv6) for outgoing SIP calls. You can also configure this feature per specific calls, using IP Profiles ('Media IP Version Preference' parameter). For a detailed description of the parameter and for configuring this feature in the IP Profiles table, see Configuring IP Profiles.

Defines the Q.850 cause value specified in the SIP Reason header that is included in a 4xx response when a Special Information Tone (SIT) is detected on an IP-to-Tel call.

The valid range is 0 to 127. The default is 34.

Note: For mapping specific SIT tones, use the following parameters: [SITQ850CauseForNC], [SITQ850CauseForIC], [SITQ850CauseForVC], and [SITQ850CauseForRO].

Defines the Q.850 cause value specified in the SIP Reason header that is included in a 4xx response when SIT-NC (No Circuit Found Special Information Tone) is detected from the Tel side for IP-to-Tel calls.

The valid range is 0 to 127. The default is 34.

Note: When not configured (i.e., default), the [SITQ850Cause] parameter is used.

Defines the Q.850 cause value specified in the SIP Reason header that is included in a 4xx response when SIT-IC (Operator Intercept Special Information Tone) is detected from the Tel for IP-to-Tel calls.

The valid range is 0 to 127. The default is -1 (not configured).

Note: When not configured (i.e., default), the [SITQ850Cause] parameter is used.

Defines the Q.850 cause value specified in the SIP Reason header that is included in a 4xx response when SIT-VC (Vacant Circuit - non-registered number Special Information Tone) is detected from the Tel for IP-to-Tel calls.

The valid range is 0 to 127. The default is -1 (not configured).

Note: When not configured (i.e., default), the {SITQ850Cause] parameter is used.

Defines the Q.850 cause value specified in the SIP Reason header that is included in a 4xx response when SIT-RO (Reorder - System Busy Special Information Tone) is detected from the Tel for IP-to-Tel calls.

The valid range is 0 to 127. The default is -1 (not configured).

Note: When not configured (i.e., default), the [SITQ850Cause] parameter is used.

Gateway application only: Assigns a Manipulation Set ID for manipulating all inbound INVITE messages.

Gateway and SBC applications: Assigns a Manipulation Set ID for manipulating incoming responses of requests that the device initiates.

The Manipulation Set is defined using the [MessageManipulations] parameter. By default, no manipulation is done (i.e. Manipulation Set ID is set to -1).

For more information, see Configuring SIP Message Manipulation.

Gateway application only: Assigns a Manipulation Set ID for manipulating all outbound INVITE messages.

Gateway and SBC applications: Assigns a Manipulation Set ID for manipulating outgoing requests that the device initiates.

The Manipulation Set is defined using the [MessageManipulations] parameter. By default, no manipulation is done (i.e. Manipulation Set ID is set to -1).

For more information, see Configuring SIP Message Manipulation.

Defines how often (in seconds) the device sends ping messages (keep alive) to check whether the WebSocket session with the Web client is still connected.

The valid value is 5 to 2000000. The default is 0 (i.e., ping messages are not sent).

For more information on WebSocket, see SIP over WebSocket.

Note:

Defines the length of each interval (in hours) in the observation window (12 intervals) for calculating average MOS of calls belonging to users registered with the device.

The valid value is 1 or 2. The default is 1.

As the device measures MOS in 12 intervals, if configured to 1, then MOS is measured over a 12 hour period; if configured to 2, then MOS is calculated over a 24 hour period. It measures the average and minimum MOS per interval. Intervals without calls are not used in the calculation.

For more information on this feature, see Configuring Voice Quality for Registered Users.

Note: This parameter is applicable only to the SBC application.

Defines the duration (in minutes) of no calls after which the MOS measurement is reset (0 and gray color). In addition, if an alternative IP Profile is configured for the Quality of Service rule and is currently being used, the device changes back to the original IP Profile.

The valid value range is 1 to 1,440. The default is 60.

For more information on this feature, see Configuring Voice Quality for Registered Users.

Note: This parameter is applicable only to the SBC application.

Defines the SIP response code that the device sends when it rejects an incoming SIP message due to a matched Message Policy in the Message Policies table, whose 'Send Reject' parameter is configured to Policy Reject.

The default is 400 "Bad Request".

To configure Message Policies, see Configuring SIP Message Policy Rules.

Defines if non-digits can be included in ENUM queries sent by the device to an ENUM server for retrieving a SIP URI address for an E.164 telephone number (destination).

For the Gateway application: ENUM queries can be used for Tel-to-IP Routing (see Configuring Tel-to-IP Routing Rules). For the SBC application: ENUM queries can be used for IP-to-IP routing with Call Setup Rules (see Configuring SBC IP-to-IP Routing Rules and Configuring Call Setup Rules).

Defines the Dial Plan that the device must search in the Dial Plans table to check if the source and destination Teams sites share a common group number. If they do, the call is a direct media call.

For more information, see Using Dial Plans for Microsoft Local Media Optimization

Note: The ini file parameter is a table, using the following syntax:

[ RegionsConnectivityDialPlan ]

FORMAT Index = RCDialPlan;

RegionsConnectivityDialPlan 0 = "NameofDialPlan";

[ \RegionsConnectivityDialPlan ]

Note: The feature is applicable only to Teams-to-PSTN calls.

Defines the device's handling of the SIP Content-Type header's value when the device sends a SIP message that has multiple bodies.

Note: The parameter is applicable only to the SBC application.

Enables the Busy Out feature.

When enabled and certain scenarios exist, the device does the following:

The above behavior is done upon one of the following scenarios:

Note:

Defines the timeout interval (in seconds) for out-of-service graceful shutdown mode for busy trunks (per trunk) if communication fails with a Proxy server (or Proxy Set). In such a scenario, the device rejects new calls from the PSTN (i.e., Serving Trunk Group), but maintains currently active calls for this user-defined timeout. Once this timeout elapses and there are still active calls, the device terminates the calls and takes the trunk out-of-service (sending the PSTN busy-out signal). Trunks without any active calls are immediately taken out-of-service regardless of the timeout.

The parameter is applicable to the locking of Trunk Groups feature (see Locking and Unlocking Trunk Groups) and the Busy Out feature (see the [EnableBusyOut] parameter), where trunks/channels are taken out-of-service.

The range is 0 to 86,400. The default is 0.

Note:

Defines which configuration table (Trunk Group Settings table or Tel-to-IP Routing table) the device uses to determine busy out for a Trunk Group.

Note:

Defines the time interval (in msec) between the first transmission of a SIP message and the first retransmission of the same message.

The default is 500.

Note: The time interval between subsequent retransmissions of the same SIP message starts with SipT1Rtx. For INVITE requests, it is multiplied by two for each new retransmitted message. For all other SIP messages, it is multiplied by two until SipT2Rtx. For example, assuming SipT1Rtx = 500 and SipT2Rtx = 4000:

Defines the maximum interval (in msec) between retransmissions of SIP messages (except for INVITE requests).

The default is 4000.

Note: The time interval between subsequent retransmissions of the same SIP message starts with SipT1Rtx and is multiplied by two until SipT2Rtx.

Defines the maximum number of UDP transmissions of SIP messages (first transmission plus retransmissions).

The range is 1 to 30. The default is 7.

Defines the number of retransmitted INVITE/REGISTER messages before the call is routed (hot swap) to another Proxy/Registrar.

The valid range is 1 to 30. The default is 3.

For example, if configured to 3 and no response is received from an IP destination, the device attempts another three times to send the call to the IP destination. If still unsuccessful, it attempts to redirect the call to another IP destination.

Note: The parameter is also used for alternative routing (see Alternative Routing Based on IP Connectivity.