Configuring SIP Servers

[Updated]

You must configure SIP Server applications for the following purposes:

• Cluster nodes
• Virtual queues
• Historical reporting

Configuring SIP Servers for SIP Cluster

[Revised]

1. Follow the standard procedure for configuring all Application objects to begin configuring these SIP Server Application objects. Deploy SIP Servers as an HA pair, Hot Standby redundancy mode.
   • Suggested application names: SIPS_<datacenter>_1, SIPS_<datacenter>_1_B
2. On the Switches tab, add the SIP Cluster Switch object to each SIP Server application.
3. On the Connections tab, add the following connections:
   • confserv_proxy_<datacenter>—Set the following parameters: Addp, Trace on both sides, Local Timeout = 7, Remote Timeout = 11
   • MessageServer_<datacenter>—Set the following parameters: Addp, Trace on both sides, Local Timeout = 7, Remote Timeout = 11
4. On the Server Info tab, configure the following ports for each SIP Server application:

ID	Listening Port	Connection Protocol
default	Any available port number
TCport	Any available port number	TController
IPport	Any available port number	IProxy

5. On the Options tab in the TServer section, configure the following mandatory options for each SIP Server application to be run in cluster mode:

Name	Cluster Value	Description
server-role	5	For SIP Server to run in cluster mode.
sip-link-type	3	For SIP Server to run in multi-threaded mode.
geo-location	String	A string identifying the data center to which this SIP Server instance belongs. It is used by SIP Proxy to select SIP Server in same data center as SIP Proxy. SIP Server uses it to select geo-location for 3PCC calls. All applications deployed in the same data center use the same value for this geo-location parameter.
sip-address		For SIP Server to build the Via and Contact headers in SIP messages.
sip-address-srv		For SIP Server to build the Via and Contact headers in SIP messages.
sip-outbound-proxy	true
sip-enable-gdns	true	To resolve SRV contacts.
sip-enable-rfc3263	true	To resolve priority and weight in SRV tables.
dial-plan		The name of the Dial Plan DN (the VoIP Service DN with service-type set to feature-server).
find-trunk-by-location	true	To enable selection of the trunk and softswitch by geo-location. This is required to keep SIP signalling on the correct data center.
enable-strict-location-match	all	For strict matching of MSML resources is required for cluster use. SIP Server in a particular geo-location must only use MCP resources in the same geo-location.
sip-enable-x-genesys-route	true	Enables a private X-Genesys-Route header in SIP messages towards SIP Proxy. It's exclusively used by and not propagated beyond the SIP Proxy.
sip-port
http-port
management-port

The sample configuration:

[agent-reservation]
request-collection-time=300 msec

[backup-sync]
addp-remote-timeout=11
addp-timeout=7
addp-trace=full
protocol=addp
[call-cleanup]
cleanup-idle-tout=60 min
notify-idle-tout=5 min
periodic-check-tout=10 min

[extrouter]
cast-type=route direct-notoken direct-callid reroute direct-uui direct-ani dnis-pool direct-digits pullback route-uui direct-network-callid

[Log]
all=/mnt/log/SIPS_VQ/SIPS_VQ
buffering=false
expire=15
segment=100 MB
spool=/mnt/log/SIPS_VQ
standard=network
time-format=iso8601
verbose=all
x-gsipstack-trace-level=3
x-server-trace-level=3

[log-filter]
default-filter-type=hide

[TServer]
acw-persistent-reasons=false
after-routing-timeout=18
agent-emu-login-on-call=true
agent-logout-on-unreg=true
agent-no-answer-action=notready
agent-no-answer-timeout=12
call-observer-with-hold=true
consult-user-data=inherited
clamp-dtmf-allowed=true
default-dn=default_rp
default-route-point=reject=404
default-route-point-order=after-dial-plan
default-music=music/on_hold_saas
dial-plan=DialPlan
divert-on-ringing=false
emulated-login-state=not-ready
extn-no-answer-timeout=12
greeting-call-type-filter=
greeting-delay-events=false
greeting-notification=
http-port=9096
init-dnis-by-ruri=true
logout-on-out-of-service=true
management-port=5002
merged-user-data=merged-over-main
monitor-consult-calls=true
msml-record-metadata-support=true
msml-record-support=true
msml-support=true
music-in-conference-file=qtmf://music/silence
override-to-on-divert=true
posn-no-answer-timeout=12
record-consult-calls=true
record-moh=false
recording-failure-alarm-timeout=900
recording-filename=$UUID$_$DATE$_$TIME$
registrar-default-timeout=140
ring-tone=qtmf://music/ring_back
rq-expire-tmout=0
rq-expire-tout=0
server-id=
set-notready-on-busy=true
shutdown-sip-reject-code=503
sip-address=<FQDN>
sip-address-srv=<SIP-SERVER-SRV>
sip-call-retain-timeout=1
sip-dtmf-send-rtp=true
sip-enable-100rel=false
sip-enable-call-info=true
sip-enable-ivr-metadata=true
sip-enable-moh=true
sip-enable-rfc3263=true
sip-interface=<FQDN>
sip-invite-treatment-timeout=15
sip-port=5060
sip-preserve-contact=true
sip-treatments-continuous=true
timeguard-reduction=1000
unknown-gateway-reject-code=503
userdata-map-trans-prefix=X-Genesys-

Configuring SIP Servers for Virtual Queues

[Updated]

[FDS: https://intranet.genesys.com/display/POD/GCloud+Voice+VM+Specification#GCloudVoiceVMSpecification-SIPServer]

Virtual Queue (VQ) SIP Servers are used primarily to manage virtual queues.

1. Follow the standard procedure for configuring all Application objects to begin configuring these SIP Server Application objects. Deploy SIP Servers as an HA pair, Hot Standby redundancy mode. One HA pair per data center. This eliminates the need to synchronize Virtual Queue states across SIP Cluster Nodes.
   • Suggested application names: SIPS_VQ, SIPS_VQ_B
2. On the Connections tab, add the following connections:
   • confserv_proxy_<datacenter>—Set the following parameters: Addp, Trace on both sides, Local Timeout = 7, Remote Timeout = 11
   • MessageServer_<datacenter>—Set the following parameters: Addp, Trace on both sides, Local Timeout = 7, Remote Timeout = 11
3. On the Switches tab, add the VQ-switch object to each VQ SIP Server application. All VQ SIP Servers must be associated with the same VQ-switch.
4. On the Server Info tab, must be only the default port.
5. On the Options tab in the TServer section, can be configured the following options:

[agent-reservation]
request-collection-time=300 msec

[backup-sync]
addp-remote-timeout=11
addp-timeout=7
addp-trace=full
protocol=addp

[call-cleanup]
cleanup-idle-tout=60 min
notify-idle-tout=5 min
periodic-check-tout=10 min

[extrouter] 
cast-type=route direct-notoken direct-callid reroute direct-uui direct-ani dnis-pool direct-digits pullback route-uui direct-network-callid

[Log]
all=/mnt/log/SIPS_VQ/SIPS_VQ
buffering=false
expire=15
segment=100 MB
spool=/mnt/log/SIPS_VQ
standard=network
time-format=iso8601
verbose=all
x-gsipstack-trace-level=3
x-server-trace-level=3

[TServer]
acw-persistent-reasons=false
after-routing-timeout=18
agent-emu-login-on-call=true
agent-logout-on-unreg=true
agent-no-answer-action=notready
agent-no-answer-timeout=12
call-observer-with-hold=true
consult-user-data=inherited
clamp-dtmf-allowed=true
default-dn=default_rp
default-route-point=reject=404
default-route-point-order=after-dial-plan
default-music=music/on_hold_saas
dial-plan=DialPlan
divert-on-ringing=false
emulated-login-state=not-ready
extn-no-answer-timeout=12
greeting-call-type-filter=
greeting-delay-events=false
greeting-notification=
http-port=9096
init-dnis-by-ruri=true
logout-on-out-of-service=true
management-port=5002
merged-user-data=merged-over-main
monitor-consult-calls=true
msml-record-metadata-support=true
msml-record-support=true
msml-support=true
music-in-conference-file=qtmf://music/silence
override-to-on-divert=true
posn-no-answer-timeout=12
record-consult-calls=true
record-moh=false
recording-failure-alarm-timeout=900
recording-filename=$UUID$_$DATE$_$TIME$
registrar-default-timeout=140
ring-tone=qtmf://music/ring_back
rq-expire-tmout=0
rq-expire-tout=0
server-id=
set-notready-on-busy=true
shutdown-sip-reject-code=503
sip-address=<FQDN>
sip-address-srv=<SIP-SERVER-SRV>
sip-call-retain-timeout=1
sip-dtmf-send-rtp=true
sip-enable-100rel=false
sip-enable-call-info=true
sip-enable-moh=true
sip-enable-rfc3263=true
sip-interface=<FQDN>
sip-invite-treatment-timeout=15
sip-port=5060
sip-preserve-contact=true
sip-treatments-continuous=true
timeguard-reduction=1000
userdata-map-trans-prefix=X-Genesys-

You will add VQ SIP Servers to the following applications:

• Stat Servers in each data center
• Each ICON that monitor SIP Servers in the same data center.
• Each URS, ORS, and SIP Feature Server located in the same data center.

Configuring SIP Servers for Historical Reporting

[Updated]

[FDS; https://intranet.genesys.com/display/POD/GCloud+SIP+Cluster+VM+Specification]

When operating in cluster mode, ICON must connect to two ports of SIP Server: T-Controller (TCport) and Interaction Proxy (IPport). For this purpose, a dummy SIP Server application must be created. When configuring ICON for historical reporting, add a connection to the IPport of the actual SIP Server application, and add a connection to the TCport of the dummy SIP Server application. Each connection represents a session. GIM requires each session to be associated with a SIP Server application.

1. Follow the standard procedure for configuring all Application objects to begin configuring this dummy SIP Server Application object.
2. On the Switches tab, add the SIP Cluster Switch, the same Switch as in the actual SIP Server application.
3. On the Server Info tab, add the same listening ports as in the actual SIP Server application. The Server Info tab must not contain HA configuration.
4. On the Connections tab, don't add anything. It must be empty.
5. On the Options tab in the [T-Server] section, don't make any changes.
6. On the Start Info tab, clear the Auto-Restart box to avoid SCS restarting the application.
7. On the Annex tab in the [sml] section, set autostart=false to avoid SCS restarting the application.

Important

The dummy SIP Server application must not be added to the applications option of the SIP Cluster DN.