I faced with a problem. I use sipml5 (html5 sip stack used from browser like chrome) and asterisk (patched with sipml5 team) but unfortunatelly when traffic goes from asterisk to chrome each srtp packet has sequence number = 0 (I catched packets with tcpdump on machine with asterisk). Also when I enable rtp debug in asterisk cli. Packets looks ok (I’m a little bit confused with packet len):
Sent RTP packet to xx.x.xx.x:58485 (via ICE) (type 08, seq 036649, ts 056288, len 4294967284)
Is it possible to get info about srtp packets (I suppose they become srtp after record above)?
Also I have to provide some additional info:
Asterisk is on public web, chrome with sipml5 is behind nat, asterisk is connected to voip provider through "register => " in sip.conf
In another setup, when all components are in same lan all things are ok, and sequence number is ok also (packet kength is also large).
Also I tried asterisk configs from another simillar asterisk setup from my friend (his setup works for him) and got the same result.
Please help me to localize a problem and solve it. If any additional information is needed I’ll provide it, just ask it.