Integrated Convergence Services

• Accounting feature—This feature is supported on SRX210 and SRX240 devices.

  You can configure Integrated Convergence Services to collect and generate accounting information for successful and unsuccessful voice subscriber transactions. The voice daemon generates and collects accounting data about calls made and received between Session Initiation Protocol (SIP), Foreign Exchange Station (FXS), and Foreign Exchange Office (FXO) stations.

  You can use the accounting feature for calls made when the SRX Series media gateway (SRX Series MGW) is in control or when the SRX Series survivable call server (SRX Series SCS) is in control.

  [Junos OS Integrated Convergence Services Configuration and Administration Guide]

• Call park—This feature is supported on SRX210 and SRX240 devices.

  The call park feature allows users to park an active call and pick up their call or that of another user later. To use the call park feature, you configure a primary logical extension, which you can think of as a parking lot. You must also configure a range of logical extensions following the primary one that are used to park individual calls.

  When a user is handling a call, they can transfer it to the parking lot without the caller hearing the transfer process. When the user parks the call, they are told the logical extension number of the parking slot before their connection to the call is dropped. That user or another one can pick up the call and resume the conversation from any phone by calling the extension number of the parking slot.

  This feature is supported when the SRX Series SCS is in control. Under normal conditions when it is reachable, the peer call server provides this service if it is supported.

  [Junos OS Integrated Convergence Services Configuration and Administration Guide]

• Defining a SIP registrar address separate from the peer call server—This feature is supported on SRX210 and SRX240 devices.

  By default, the SIP registrar and the peer call server (SIP server) are handled by the same service and therefore have the same address. Under these circumstances, the SRX Series MGW sends SIP REGISTRAR and INVITE messages to the IP address configured for the peer call server.

  In some SIP network environments, the registrar and the peer call server are separate entities. For these network environments, you can specify separate addresses for the registrar and the peer call server. If you configure a separate address for the registrar, INVITE messages are sent to the peer call server and REGISTER messages are sent to the registrar. If you do not configure a registrar address, the default behavior takes effect.

  [Junos OS Integrated Convergence Services Configuration and Administration Guide]

• Defining a SIP registrar address separate from the peer proxy server—This feature is supported on SRX210 and SRX240 devices.

  By default, the SIP registrar and the peer proxy server (SIP server) are handled by the same service and therefore have the same address. In this case, the SRX Series MGW sends SIP REGISTRAR and INVITE messages to the IP address configured for the peer proxy server.

  In some SIP network environments, the registrar and the peer proxy server are separate entities. For these network environments, you can specify separate addresses for the registrar and the peer proxy server. If you configure a separate registrar address, INVITE messages are sent to the peer proxy server and REGISTER messages are sent to the registrar. If you do not configure a registrar address, the default behavior takes effect.

  [Junos OS Integrated Convergence Services Configuration and Administration Guide]

• Direct inward dialing lists—This feature is supported on SRX210 and SRX240 devices.

  You can associate a list of direct inward dialing (DID) numbers with a trunk to be used for assignment to stations. You do not need to assign these DIDs to stations directly. The software assigns a DID number to a single station exclusively. If an incoming call is made to an unassigned DID number, it is directed to and handled by auto attendant.

  [Junos OS Integrated Convergence Services Configuration and Administration Guide]

• Disabling SIP registration to the peer call server—This feature is supported on SRX210 and SRX240 devices.

  The SRX Series MGW sends registration messages to the peer call server. For some network environments in which all media gateways are known to the peer call server, the SRX Series MGW is not required to register to it. To do so could cause complications. For example, the peer call server could drop the registration message “silently,” that is, without informing the SRX Series MGW. In this case, the SRX Series MGW might retransmit the message, incurring unnecessary processing and adding to the network load.

  When you configure peer call server information, you can disable transmission of the registration message to the peer call server to avoid these problems

  Note: Disabling transmission of the SRX Series MGW registration to the peer call server does not disable registration of an FXS station to the SRX Series MGW on the device running Integrated Convergence Services.

  [Junos OS Integrated Convergence Services Configuration and Administration Guide]

• Disabling SIP registration to the proxy server—This feature is supported on SRX210 and SRX240 devices.

  By default, Integrated Convergence Services SIP trunks register to the SIP service provider’s peer proxy server. For some SIP networks, the peer proxy server is informed about all SIP trunks that communicate with it. In such network environments, the SIP trunk does not need to send a REGISTER message to the peer proxy server. To do so would increase network load unnecessarily. To accommodate these network environments, you can configure the SIP trunk not to register to the peer proxy server.

  [Junos OS Integrated Convergence Services Configuration and Administration Guide]

• DSCP marking for RTP packets generated by SRX Series Integrated Convergence Services—This feature is supported on SRX210 and SRX240 devices that have high memory, power over Ethernet capability, and media gateway capability.

  Configure DSCP marking to set the desired DSCP bits for RTP packets generated by SRX Series Integrated Convergence Services.

  DSCP bits are the 6-bit bitmap in the IP header used by devices to decide the forwarding priority of packet routing. When the DSCP bits of RTP packets generated by Integrated Convergence Services are configured, the downstream device can then classify the RTP packets and direct them to a higher priority queue in order to achieve better voice quality when packet traffic is congested. Juniper Networks devices provide classification, priority queuing, and other kinds of CoS configuration under the Class-of-Service configuration hierarchy.

  Note that the Integrated Convergence Services DSCP marking feature marks only RTP packets of calls that it terminates, which include calls to peer call servers and to peer proxy servers that provide SIP trunks. If a call is not terminated by Integrated Convergence Services, then DSCP marking does not apply.

  To configure the DSCP marking bitmap for calls terminated by Integrated Convergence Services and the address of the peer call server or peer proxy server to which these calls are routed, use the media-policy statement in the [edit services converged-services] hierarchy level.
  set services convergence-service service-class < name > dscp < bitmap >
  set services convergence-service service-class media-policy < name > term < term-name > from peer-address [< addresses >]
  set services convergence-service service-class media-policy < name > term then service-class < name >

  [Junos OS Integrated Convergence Services Configuration and Administration Guide]

• Hunt group—This feature is supported on SRX210 and SRX240 devices.

  A hunt group enables a group of users to handle calls collectively. A hunt group specifies a logical extension that outside parties can call. Member stations belonging to the hunt group are specified in a preconfigured station group. When a call comes in on the logical extension, the call is directed to the phone whose station is specified first in the preconfigured station group, and that phone rings. The next incoming call is directed to the second station specified in the station group and its phone rings, and so on.

  To connect the call, the system hunts through the configured stations in order one at a time. It rings a phone up to the time limit that you specify before it tries the next phone in the configured order

  This feature is supported when the SRX Series SCS is in control. Under normal conditions when it is reachable, the peer call server provides this service if it is supported.

  [Junos OS Integrated Convergence Services Configuration and Administration Guide]

• Pickup group—This feature is supported on SRX210 and SRX240 devices.

  Pickup groups enable users to handle incoming calls collectively, as a group. Members of the same pickup group can answer incoming calls directed at any phone extension number within the group. When a phone is called, the first available agent takes the call, whether it comes in on their phone or another phone within the group. To pick up a call, the user dials the digits *8. After the user takes the call, the phone whose number was called no longer rings. Users can belong to one or more pickup groups concurrently.

  The pickup group feature rings only one phone at a time. If the first phone tried is busy, the next one is tried, and so on. A pickup group can include up to 20 members whose phones can be either analog or SIP, but not a mix of both.

  This feature is supported when the SRX Series survivable call server (SRX Series SCS) is in control. Under normal conditions when it is reachable, the peer call server provides this service if it is supported.

  [Junos OS Integrated Convergence Services Configuration and Administration Guide]

• Ring group—This feature is supported on SRX210 and SRX240 devices.

  A ring group can include up to five members. A ring allows incoming calls to be handled by any member of the group. You configure a ring group with a logical extension that outside parties can call. Calls coming into the logical extension are forwarded to all phones simultaneously. The first member to answer the call takes it, and the phones of other members of the group stop ringing. A ring group can include both SIP and analog stations.

  This feature is supported when the SRX Series SCS is in control. Under normal conditions when it is reachable, the peer call server provides this service if it is supported.

  [Junos OS Integrated Convergence Services Configuration and Administration Guide]