# SIP networking

You can network several communication servers via SIP. You can set up the SIP nodes here.

## Settings for the remote SIP nodes

Table 1. General settings
Parameter | Remarks
SIP node | Reference number of the SIP node
Name | Name of the SIP node
Bandwidth area | The codecs used are specified with the assigned bandwidth area.
Presence information mode | No music: The remote SIP node does not support presence information. Server: The remote SIP node is a presence information server (e g. a BluStar Server). Peer: The remote SIP node is a presence information peer (e.g. an MiVoice Office 400 communication server or an Mitel Mobile Client Controller). Note: Set the parameter to None, if the remote SIP node does not provide presence information. This helps reduce the traffic load.
Trunk group | Assign a trunk group to the remote SIP node.

Table 2. IP addressing
Parameter | Remarks
IP address: IP port | Enter the IP address and the IP port of the remote SIP node here.

Table 3. SIP signalling
Parameter | Explanation
Use ‘+’ for the international prefix | The call number is also included, in canonical format.
Try to make external calls: Timeout | Once that time has elapsed, the communication server tries to set up the call via the next trunk group defined in the route (default value: 32 seconds).
’From’ field for CLIR | If call identification is suppressed at the calling user, the following sender is provided, depending on the selection (display name and address): Anonymous with privacy/critical ( 3261): Display name: anonymous@anonymous.invalid; Address: anonymous@anonymous.invalid As defined in SIP account (RFC 3323): Display name and address as defined in the SIP account. Displayed name is ’Anonymous’: Display name: anonymous@anonymous.invalid; address remains unchanged. Anonymous without privacy header (RFC 3261): Display name: anonymous@anonymous.invalid; Address: anonymous@anonymous.invalid
Send session refresh (RFC 4026) | The communication server attempts to negotiate a period for regular "Session Refresh Messages" with the remote SIP node. For this, the remote SIP node must support RFC4028.
Use destination URL from | The destination URL can be formed from the ’To’-Field or from the request line. The choice depends on the remote SIP node.
Music on hold | Music on hold is played, provided it is activated throughout the system.
Music on hold: Signalling | The type of signalling for music on hold depends on what the SIP provider supports: Automatic: The communication server itself tries to recognise which of the two RFCs the SIP provider supports. According to RFC 3264 The SIP provider supports signalling according to the RFC An Offer/Answer Model with the Session Description Protocol, (SDP), June 2002. According to RFC 2543 The SIP provider supports signalling according to the RFC SIP: Session Initiation Protocol As active media update: The communication server keeps the 2-way media connection. This allows to play music on hold into the call channel from the communication server instead from the SIP provider. Signal connection update: The SIP provider is informed about the change of the media port by a separate SDP message. No signalling (no media update): the SIP provider does not deliver any signalling and the communication server plays music on hold into the call channel.
Send redirecting information | Redirecting information allows the called party to see whether the call was redirected and, if so, by whom. Two different methods are defined for this purpose for SIP. The parameter can be configured for each SIP node as well as for each SIP terminal. No: No redirection/redirecting information is displayed. Yes, using 'Diversion header (recursing)': Redirection/redirecting information is displayed in the case of incoming calls only. The call is forwarded in the communication server. Yes, using 'Diversion header (non recursing)': The call forwarding for outgoing calls is indirect, with the communication server sending back ‘Response 302’ (Moved Temporarily) with the necessary redirection information to the SIP phone. The SIP phone itself then makes the call to the forwarding destination and shows the redirection/redirecting information on its own display. Note: Call forwarding with Response 302 is not possible in every case.
Call transfer mode | You can select here whether the REFER or Re-INVITE method should be used during an external call transfer. Note: The REFER method is only used when both of the users to be transferred are found on the same SIP node.
PRACK support (RFC 3262) | The PRACK method according to RFC 3262 is supported.

Table 4. Audio settings
Parameter | Remarks
Preferred codec | Choose the preferred codec here: Unspecified: A suitable codec will be set automatically. G.711a: Uncompressed codec with high audio quality. Suitable for links with large bandwidths. The bit rate is 64 kbit/s. Uses the German tone signalling process. G.711u: Uncompressed codec with higher audio quality. Suitable for links with large bandwidths. The bit rate is 64 kbit/s. Uses the American tone signalling process. G.729: Compressed codec with medium audio quality. Suitable for links with limited bandwidth. The bit rate is 8 kbit/s.
Comfort noise support | Off: Passive: Active:
RTCP support | Off: Passive: Active:

Table 5. NAT settings
Parameter | Remarks
Enable keep alive | The communication server periodically updates the NAT table on its own firewall using notification messages to the proxy server. This means that the system remains reachable for incoming SIP calls.
ALG support | Supports the connection to the remote SIP node. The IP packets that contain SIP signalling information are opened by the ALG (Application Layer Gateway) and the private IP address is replaced by the public IP address. (The public IP address in the communication server must be configured.)
Relay RTP data via communication server | Indirect switching: During connection set-up to another IP endpoint, the voice data is forwarded via the communication server and not directly. This can help in solving NAT and firewall problems. Direct switching: During connection set-up to another IP endpoint, the voice data is forwarded directly. Note: Indirect switching requires two additional VoIP channels on the communication server.

Table 6. Authentication and transport protocol
Parameter | Remarks
Local authentication required | Authentication on the remote SIP node is required.
User name | Enter the user name for authentication on the remote SIP node here.
Password | Enter the password for authentication on the remote SIP node here.
Transport protocol | Here you can select the transport protocol you want.

Parent topic: Private networking
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