iPECS UCP - Keep Alive; NAT Resolution; Session Timer; Srtp; System Firewall Resolution

iPECS eMG80 & eMG800 & UCP
Administration and Programming Manual Issue 1.6
465
authentication, user login Password should be available in PGM 443 for the
Station.
< Keep Alive / NAT Resolution >
To keep stable information of SIP Phone‘s Connection, IP address and Port number that is under
NAT environment, system uses ‘OPTIONS’ message to implement Keep Alive and assist NAT
resolution - effort to maintain IP address of SIP Phone by sending message so often from system
to SIP Phone. SIP Phone should be capable to answer for ‘OPTIONS’ message
− Check Message Sending Timer in [SIP Data / SIP Attributes (210)] : 120 seconds
− Keep Alive Usage for a SIP Station in [SIP Data / SIP Phone Attributes (211)] : ON
− Retry Count for a SIP Station in [SIP Data / SIP Phone Attributes (211)] : 3
< System Firewall Resolution >
In case of firewall routed with MPB, to distinguish remote SIP Phone that is outside of firewall
from system local area a check bit is required per a SIP Station. With this check bit, system can
determine whether to serve communication using firewall mapped WAN IP address of MPB or
serve communication using LAN IP address of MPB/UCP.
− SIP Phones that are outside of system protect firewall : [SIP Data / SIP Phone
Attributes (211)] – ‘Same Firewall with MPB/UCP’ to ‘OFF’
< Session Timer >
To confirm talk state frequently during in talk state, system sends ‘UPDATE’ message to SIP
Phone. If there is no response for the UPDATE message with in Maximum session timer, system
will disconnect the talking call.
− [SIP Data / SIP Phone Attributes (211)] – Session Timer Support : ON
− [SIP Data / SIP Phone Attributes (211)] – Max Session Timer : if exceed, disconnect
talking call
− [SIP Data / SIP Phone Attributes (211)] – Min Session Timer: minimum guard timer for
session timer negotiation.
< SRTP >
Voice & Video Data Encryption requires synchronization of CRYPTO method between system
and SIP Phone side. If system specifies SRTP information then same information should be in
SIP Phone side by Phone user programming.
SRTP usage requires a SRTP relay channel via system VOIU and VOIB/VOIM.
− [SIP Data / SIP Phone Attributes (211)] – SRTP Usage: ON
→SIP Phone self-programming is required, too – SRTP ON
− [SIP Data / SIP Phone Attributes (211)] – 1st CRYPTO key generation type: one of
ARIA_CM_192_HMAC_SHA1_80, AES_CM_128_HMAC_SHA1_80,
ARIA_CM_128_HMAC_SHA1_80
→SIP Phone self-programming is required, too – 1st/2nd CRYPTO method
− [SIP Data / SIP Phone Attributes (211)] – 2nd CRYPTO key generation type: one of
ARIA_CM_192_HMAC_SHA1_80, AES_CM_128_HMAC_SHA1_80,
ARIA_CM_128_HMAC_SHA1_80
→SIP Phone self-programming is required, too – 1st/2nd CRYPTO method