Audio stream sequence warning

[mgsray]

I use sipsorcery to call remote freeswitch server, it works fine. But sometimes the log shows the warning of Audio stream sequence number jumped. In some cases, it keeps reporting this warning for 10~20 times in almost every seconds.
What's the meaning, and does it have any impact on the application?

[13:10:24 DBG] Response 200 OK for sip:CUSTOMER-VP-391@fs01.xxx.com:15060.
[13:10:24 DBG] Setting audio sink and source format to 0:PCMU 8000 (RTP clock rate 8000).
[13:10:24 INF] Call attempt to sip:CUSTOMER-VP-391@fs01.xxx.com:15060 was answered.
[13:10:25 DBG] Set remote audio track SSRC to 2449649697.
[13:10:56 WRN] Audio stream sequence number jumped from 15685 to 15687.
[13:10:58 DBG] Request received: udp:0.0.0.0:61226<-udp:139.XXX.XXX.XXX:15060 BYE sip:192_168_20_1-391-232@172.16.X.30:61226 SIP/2.0
[13:10:58 INF] Remote call party hungup BYE sip:192_168_20_1-391-232@172.16.X.30:61226 SIP/2.0.

[sipsorcery]

It means the RTP packets from the remote party are not getting through to your application, i.e. packet loss. A small amount of loss may not be noticeable but if it gets too large you'll notice audio chop or breaks. Generally the cause is network congestion but if you notice a pattern to it then it could be your FreeSWITCH server is overloaded, your application is CPU bound or some other condition.