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SIP UDP packet loss?

2011-03-19 06:13:34
Hallo  Sip Experts / Transport Group Experts,
I am not a SIP expert of application layer. I am part of Test System
Development where we simulate the SIP traffic over different transport types
like UDP,TCP and SCTP.
When we are simulating just SIP signalling messages between our two test
systems at the rate of  greater than 500  UDP packets per second, we
observed some of
Subscribers fail Signalling.After further investigations it is found that
UDP packets are lost  consequently Signalling is failing for
some subscribers intermittently.

To overcome this we then increased the udp.recvspace and udp.sendspace and
socketbuffer size in our Networking stack.
This helped the packet loss to reduce by 70% but could not achieve 0% packet
loss .
*Qeury 1*:- Is increasing buffer space is a solution or workaround for our
Network stack  to overcome packet loss ?

Default values of our Network stack  were 40K for udp.recvspace and 9K for
udp.sendspace and socketbuffers are 256K.
We changed these values in our Network stack  to 256K  for udp.recvspace,
256K for udp.sendspace and socketbuffers to 1024K.
This change has reduced packet loss but can could not make 0% packet loss.

Query2:- Can you please suggest us what should be the right values for such
buffer space if this is right approach ?

We have yet to enable the voice transmission as a next immediate step, then
again SIP RTP UDP packets will further increase based on the codec chosen
and this will
cause even more packet loss.

Query3:-  What is the ultimate  solution for eliminating packet loss both on
Signalling and on speech path ?

Thank you in advance for your time.
Samba.
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