iPECS eMG80/100& eMG800 & UCP & vUCP
Administration and Programming Manual Issue 2.5
274
Keep Alive / NAT Resolution
To keep stable information of SIP Phone‘s Connection, IP address, and port number that is the under NAT
environment, the system uses ‘OPTIONS’ message to implement Keep-Alive and assist NAT resolution - effort
to maintain the IP address of SIP Phone by sending a 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 the case of a firewall routed with MPB, a check bit is required per a SIP Station to distinguish remote SIP
Phones outside the firewall from the local system.
With this check bit, the 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 outside the system protect firewall: [SIP Data / SIP Phone Attributes (211)] – ‘Same
Firewall with MPB/UCP’ to ‘OFF.’
Session Timer
To confirms the talk state frequently during the talk state, the system sends an ‘UPDATE’ message to SIP
Phone. If there is no response for the UPDATE message within the Maximum session timer, the 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 it exceeds, 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 the system specifies SRTP information, then the same information should be in SIP Phone side by
Phone user programming.
SRTP usage requires an 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
To Next Page
Loading...
