There could be bottle neck in your application. Maybe it's unable to read all the UDP packets from the network. You need to find out what is the capacity of your application.
-----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of Sambasiva Rao MANCHILI Sent: Sunday, March 20, 2011 2:31 PM To: [email protected] Cc: [email protected] Subject: [Sip-implementors] SIP UDP packet loss? Hallo Sip Implementors, 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 10%. Before increasing buffers we have Faults rate of ~8 to 10% on Signalling after increasing buffers we have fault rate of 0.9% over Signalling. *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. I understand UDP is not reliable transport, but I am looking for solution which can reduce Packet loss to less than 1%. When I enabled speech path codec G.729/ G.711 I see still more packet loss of 15%. 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. _______________________________________________ Sip-implementors mailing list [email protected] https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors _______________________________________________ Sip-implementors mailing list [email protected] https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
