the correct scheme is PBX---E1--Cisco 2811-----VoIP SIP-----FreeSWITCH-----VoIP SIP----NAT----asterisk(1.8.20)---IP Phone Asterisk IP is 162.48.13.26 (External IP NAT ) 30.10.2013 22:36, "Гришин В.С." пишет:
the scheme is PBX---E1--Cisco 2811-----VoIP SIP-----FreeSWITCH-----VoIP SIP---asterisk(1.8.20)---IP Phone
FreeSWITCH IP is 191.21.19.12 Asterisk IP is 162.48.13.26 Call comes to FreeSWITCH from a Cisco 2811.
SIP session between freeswitch and asterisk(1.8.20)
|Time | 191.21.19.12 | | | | 162.48.13.26 | |21.048754| INVITE SDP (g711A CN) |SIP From: "91295" <sip:9195@191.21.19.12 To:<sip:9647@162.48.13.26 | |(5065) ------------------> (5060) | |21.048758| INVITE SDP (g711A CN) |SIP From: "91295" <sip:9195@191.21.19.12 To:<sip:9647@162.48.13.26 | |(5065) ------------------> (5060) | |21.065794| 100 Trying| |SIP Status | |(5065) <------------------ (5060) | |21.122406| 180 Ringing |SIP Status | |(5065) <------------------ (5060) | |21.143580| RTP (g711A) |RTP Num packets:50 Duration:0.922s SSRC:0xB5320A6 | |(21866) <------------------ (12772) | |22.282684| 180 Ringing |SIP Status | |(5065) <------------------ (5060) | |22.296762| 181 Call is being forwarded |SIP Status | |(5065) <------------------ (5060) | |27.558268| 180 Ringing |SIP Status | |(5065) <------------------ (5060) | |35.039883| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |35.307222| RTP (g711A) |RTP Num packets:1490 Duration:29.773s SSRC:0x530CAB37 | |(21866) <------------------ (12772) | |35.476614| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |35.532995| RTP (g711A) |RTP Num packets:2966 Duration:29.680s SSRC:0x308FC816 | |(21866) ------------------> (12772) | |36.436846| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |38.282686| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |42.025399| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |45.637280| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |49.343115| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |53.119940| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |56.907800| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |60.791453| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |64.606730| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |65.219392| BYE | |SIP Request | |(5065) <------------------ (5060) | |65.234354| 200 OK | |SIP Status | |(5065) ------------------> (5060) | |65.234359| 200 OK | |SIP Status | |(5065) ------------------> (5060) |
It is visible that there is no package with ACK confirmation from a FreeSWITCH to Asterisk. Asterisk sends several times SIP OK on a freeswitch, and without having waited confirmations (ACK) after 23 seconds sends SIP BY on a freeswitch and on IP Phone. After that call stops. Before disconnection subscribers hear each other of 29 (+-) seconds
In tcpdump between Cisco and FreeSWITCH it is visible that the SIP ACK packet from Cisco arrives on a freeswitch.
In what place the SIP ACK in a FreeSWITCH disappeared? What to me to analyse in SIP session? Maybe someone solved a similar problem? please, Help!
Vladislav Grishin
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Vladislav Grishin