Technical data

8 ServerIron ADX Advanced Server Load Balancing Guide
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SIP SLB and Call Persistence using ServerIron ADX
1
FIGURE 3 SIP server farm with DSR mode
Figure 3 demonstrates a typical use case - the ServerIron application switch provides call-id based
server persistence for UDP SIP traffic. The Call-ID attribute that uniquely identifies a SIP call is used
to maintain session persistence. Due to the unique call flow requirements of SIP, most SIP
implementations require you to enable direct server return (DSR) mode on the ServerIron switch.
Since User1's SIP phone does not know the location of User2's SIP phone, it initiates a new SIP
session by sending INVITE request to SIP Proxy server. It also generates a unique identifier (Call-ID)
for the call. Because the SIP proxy server used by User1's SIP phone is actually the virtual IP
address hosted on the ServerIron switch, the ServerIron switch receives the INVITE request and,
using a server selection mechanism, identifies the best available SIP server for this INVITE. The
ServerIron uses the call-ID attribute value to select one of SIP servers in either stateless or stateful
mode. For all SIP transactions within a dialog that use same call-ID, the ServerIron selects the
same SIP server. A new INVITE message with a different call-ID is again subjected to Server Load
Balancing and may be forwarded to a different SIP server.
The proxy server receives the INVITE request and sends 100 (Trying) message to User1's SIP
phone. Since the ServerIron switch is configured in DSR mode, the response message that is
sourced from the virtual IP address flows directly to User1's SIP phone, bypassing the ServerIron.
The proxy server then consults the location service and forwards the INVITE request directly to
User2's SIP phone, again bypassing the ServerIron and is sourced from the proxy server's own IP
address.
INVITE F1
INVITE F1
ACK F8
MEDIA RTP
BYE F9
OK F10
TRYING F3
RINGING F5
OK F7
INVITE F2
ServerIron
SIP Proxy Server
user2
user1
From/To SIP Phone To/From VIP
L2/3
Infrastructure
RINGING F4
OK F6
From/To SIP Phone To/From Real IP
SI