Configuring Signaling Protocol Parameters

Note: In the current version release, only SIP (Session Initiation Protocol) is supported.

¾ To configure signaling protocol parameters:

„ After clicking the menu 'Voice over IP' in the main screen, the 'Signaling Protocol' screen opens by default (refer to the figure below).

Figure 5-1: VoIP - Signaling Protocol

Table 5-1: VoIP - Signaling Protocol

Parameter Description

Use SIP Proxy When checked, outgoing calls will be routed to the configured SIP proxy. If the parameter 'Use SIP Proxy IP and Port for Registration' is checked as well, the configured SIP proxy will also be used as the registrar, allowing incoming calls.

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„ Check the checkbox 'Use SIP Proxy'; the 'SIP Proxy' screen (showing basic

parameters) opens. Click the button 'Advanced>>'; the 'SIP Proxy' screen (showing the advanced parameters, including the basic parameters) opens.

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Table 5-2: VoIP - Signaling Protocol - Signaling Protocol

Parameter Description

SIP Transport Protocol* Choose either UDP (default) or TCP.

Local SIP Port* The UDP/TCP port (default = 5060) on which the Stack listens.

Gateway Name-User

Domain* This domain name will be sent in the From header of outgoing Invite messages.

Enable PRACK* When enabled, the MP-202 replies with a PRACK message upon receipt of a reliable provisional response.

The MP-202 does not initiate reliable provisional responses.

Include ptime in SDP* When enabled, the MP-202 adds the ptime field to the SDP message body.

Enable rport* When enabled, the MP-202 adds the rport parameter to the relevant SIP Message fields.

Connect media on 180* When enabled, media is connected upon receipt of SIP 180, 183, or 200 messages. When the parameter is disabled, media is connected upon receipt of 183 and 200 messages only.

Enable Keep Alive

using OPTIONS* When enabled, a SIP OPTIONS message is sent periodically to the SIP registrar entity.

Keep alive period** Sets the periodic interval.

* This parameter appears only in 'Advanced' mode.

* * This parameter appears only in 'Advanced' mode and when “Enable Keep Alive using OPTIONS” is enabled.

Table 5-3: VoIP - Signaling Protocol - SIP Proxy and Registrar

Parameter Description

Use SIP Proxy When checked, outgoing calls will be routed to the configured SIP proxy. If the parameter 'Use SIP Proxy IP and Port for Registration' is checked as well, the configured SIP proxy will also be used as the registrar, allowing incoming calls.

Proxy IP Address or

Host Name The IP address or host name of the SIP proxy. Proxy Port The UDP or TCP port of the SIP proxy.

Maximum Number of

Authentication Retries Defines how many times authenticated register messages are re-sent if 401 or 407 responses with a different “nonce” are received.

SIP Security The MP-202's firewall can be configured to block incoming packets that have the SIP signaling port as their destination. You can configure up to two SIP entities (for example, the SIP Proxy or an SBC), which are not to be blocked by the firewall.

The default value is “Allow all SIP traffic”.

Address Type** Defines the address type of the additional SIP entity. It can be set to “IP Address” or “Host Name”.

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Parameter Description

Use SIP Proxy IP and

Port for Registration Use the SIP proxy IP and port for registration. Default = checked. When checked, there is no need to configure the address of the registrar separately.

Use SIP Outbound

Proxy* Use an outbound SIP proxy (all SIP messages will be sent to this server as the first hop). Default = unchecked.

* This parameter appears only in 'Advanced' mode.

** This parameter appears only if the parameter 'SIP Security' is set to “Allow SIP traffic from Proxy and Additional SIP Entity”.

Table 5-4: VoIP - Signaling Protocol - SIP Timers

Parameter Description

Retransmission Timer

T1 The SIP T1 retransmission timer according to RFC 3261 Retransmission Timer

T2

The SIP T2 retransmission timer according to RFC 3261

Retransmission Timer

T4 The SIP T4 retransmission timer according to RFC 3261 INVITE Timer The SIP INVITE timer according to RFC 3261

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„ Uncheck the box 'Use SIP Proxy IP and Port for Registration' and check 'Use SIP Registrar'; the parameters screen for 'SIP Registrar' opens (showing the basic parameters).

Figure 5-3: VoIP - Signaling Protocol - SIP Proxy and Registrar

The table below shows descriptions of those SIP Registrar parameters that differ from SIP Proxy parameters. Descriptions of common parameters can be seen under the section 'SIP Proxy and Registrar', above.

Table 5-5: VoIP - Signaling Protocol - SIP Registrar SIP

Parameter Description

Use SIP Registrar Check the box to use a separate SIP registrar server.

Registrar Address The IP address or host name of the registrar server.

Registrar Port The UDP or TCP port of the registrar server.

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Click the button 'Advanced', and then check the box 'Use SIP Outbound Proxy'; the parameters screen for 'SIP Outbound Proxy' opens (showing the advanced parameters, including the basic parameters).

Figure 5-4: VoIP - Signaling Protocol - SIP Outbound Proxy

Table 5-6: VoIP - Signaling Protocol - SIP Outbound Proxy

Parameter _Description

Outbound Proxy IP The IP address of the outbound Proxy. If this parameter is set, all outgoing messages (including Registration messages) will be sent to this Proxy according to the Stack behavior.

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Click the button 'Advanced', and then check the box 'Enable STUN'; the parameters screen for 'NAT Traversal' opens (showing the advanced parameters, including the basic parameters).

Figure 5-5: VoIP - Signaling Protocol - NAT Traversal

Table 5-7: VoIP - Signaling Protocol - NAT Traversal

Parameter _Description

Enable STUN When checked, the SIP STUN Manager starts. SIP STUN Manager resolves private addresses that need to be resolved to public addresses.

STUN Server Address* The IP address of the STUN server used to resolve private addresses.

STUN Server Port* The port of the STUN server.

Subnet Mask* The subnet mask address of the STUN server used to resolve private addresses.

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