Hi, can i please play back a .pcap file ? because when i converted this file .pcap to new file .wav the codec is changed ,can you help me please ?? best regards amani
The codec doesn't magically change itself. If you want to change the codec you can use sox to do so, if you don't, you don't have to either. What exactly are you trying to accomplish here? On Thursday, March 31, 2016, amani mansour <amani.mansour2@gmail.com> wrote:
Hi,
can i please play back a .pcap file ? because when i converted this file .pcap to new file .wav the codec is changed ,can you help me please ??
best regards amani
sir i will call a FS(extension 600) from my sip phone(extension 423) i used a gateway when the call is ringing i will hear a short msg , this msg is a pcap file ,when i changed it to wav file automatically in freeswitch it has changed to the first codec selected in my soft phone : my pcap file has us codec g711 my phoner lite i have selected 1/opus 2/ g711A 3./......... but when i play this wav i have in tshark this line: 8533 960.577294 192.168.3.5 -> 192.168.3.1 RTP 109 PT=opus, SSRC=0x57A054EC, Seq=2363, Time=424320 my dialplan/default.xml: <extension name="600"> <condition field="caller_id_number" expression="^(10[01][0-9]|1500|423|400|1011)$"> <action application="set" data="continue_on_fail=true"/> <action application="ring_ready"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="ignore_early_media=false"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="[leg_timeout=20,absolute_codec_string=PCMA ]sofia/gateway/gateway_to_600/600"/> <action application="answer"/> </condition> </extension> <extension name="600"> <condition field="destination_number" expression="^(600)$"> <action application="export" data="nolocal:codec_string=PCMA"/> <action application="answer"/> <action application="info"/> <action application="sleep" data="1000"/> <action application="playback" data="/home/amani/projet/PFE/gg711.pcap.0xd2bd4e3e.wav"/> </condition> </extension> i have us tcpdump : 14:59:31.659246 IP 192.168.3.1.sip > 192.168.3.5.sip: SIP: INVITE sip:600@192.168.3.5 SIP/2.0 14:59:31.659773 IP 192.168.3.5.sip > 192.168.3.1.sip: SIP: SIP/2.0 100 Trying 14:59:31.661679 IP 192.168.3.5.sip > 192.168.3.1.sip: SIP: SIP/2.0 407 Proxy Authentication Required 14:59:31.662289 IP 192.168.3.1.sip > 192.168.3.5.sip: SIP: ACK sip:600@192.168.3.5 SIP/2.0 14:59:31.662726 IP 192.168.3.1.sip > 192.168.3.5.sip: SIP: INVITE sip:600@192.168.3.5 SIP/2.0 14:59:31.663177 IP 192.168.3.5.sip > 192.168.3.1.sip: SIP: SIP/2.0 100 Trying 14:59:38.508070 IP 192.168.3.5.sip > 192.168.3.1.35200: Flags [.], ack 2002, win 789, length 0 14:59:38.508188 IP 192.168.3.1.35200 > 192.168.3.5.sip: Flags [.], ack 2260, win 253, length 0 14:59:41.823972 IP 192.168.3.5.sip > 192.168.3.1.sip: SIP: SIP/2.0 180 Ringing 14:59:52.073381 IP 192.168.3.5.sip > 192.168.3.1.sip: SIP: SIP/2.0 183 Session Progress 15:00:02.027890 IP 192.168.3.5.sip > 192.168.3.1.sip: SIP: SIP/2.0 200 OK 15:00:02.028681 IP 192.168.3.1.sip > 192.168.3.5.sip: SIP: ACK sip:600@192.168.3.5:5060;transport=udp SIP/2.0 15:00:02.034884 IP 192.168.3.5.sip > 192.168.3.1.sip: SIP: BYE sip:423@192.168.3.1:5060 SIP/2.0 15:00:02.035367 IP 192.168.3.1.sip > 192.168.3.5.sip: SIP: SIP/2.0 200 OK my supervisor tel me that when i change the file to .wav nrmally FS code and decode the file using unknown codec can you help me please sir ??? best regards amani
https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation <https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation>
On Mar 31, 2016, at 10:31 AM, amani mansour <amani.mansour2@gmail.com> wrote:
sir i will call a FS(extension 600) from my sip phone(extension 423) i used a gateway when the call is ringing i will hear a short msg , this msg is a pcap file ,when i changed it to wav file automatically in freeswitch it has changed to the first codec selected in my soft phone :
my pcap file has us codec g711 my phoner lite i have selected 1/opus 2/ g711A 3./.........
but when i play this wav i have in tshark this line: 8533 960.577294 192.168.3.5 -> 192.168.3.1 RTP 109 PT=opus, SSRC=0x57A054EC, Seq=2363, Time=424320
my dialplan/default.xml: <extension name="600"> <condition field="caller_id_number" expression="^(10[01][0-9]|1500|423|400|1011)$"> <action application="set" data="continue_on_fail=true"/> <action application="ring_ready"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="ignore_early_media=false"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="[leg_timeout=20,absolute_codec_string=PCMA ]sofia/gateway/gateway_to_600/600"/> <action application="answer"/>
</condition> </extension>
<extension name="600"> <condition field="destination_number" expression="^(600)$"> <action application="export" data="nolocal:codec_string=PCMA"/> <action application="answer"/> <action application="info"/> <action application="sleep" data="1000"/> <action application="playback" data="/home/amani/projet/PFE/gg711.pcap.0xd2bd4e3e.wav"/> </condition> </extension>
i have us tcpdump : 14:59:31.659246 IP 192.168.3.1.sip > 192.168.3.5.sip: SIP: INVITE sip:600@192.168.3.5 <mailto:sip%3A600@192.168.3.5> SIP/2.0 14:59:31.659773 IP 192.168.3.5.sip > 192.168.3.1.sip: SIP: SIP/2.0 100 Trying 14:59:31.661679 IP 192.168.3.5.sip > 192.168.3.1.sip: SIP: SIP/2.0 407 Proxy Authentication Required 14:59:31.662289 IP 192.168.3.1.sip > 192.168.3.5.sip: SIP: ACK sip:600@192.168.3.5 <mailto:sip%3A600@192.168.3.5> SIP/2.0 14:59:31.662726 IP 192.168.3.1.sip > 192.168.3.5.sip: SIP: INVITE sip:600@192.168.3.5 <mailto:sip%3A600@192.168.3.5> SIP/2.0 14:59:31.663177 IP 192.168.3.5.sip > 192.168.3.1.sip: SIP: SIP/2.0 100 Trying 14:59:38.508070 IP 192.168.3.5.sip > 192.168.3.1.35200: Flags [.], ack 2002, win 789, length 0 14:59:38.508188 IP 192.168.3.1.35200 > 192.168.3.5.sip: Flags [.], ack 2260, win 253, length 0 14:59:41.823972 IP 192.168.3.5.sip > 192.168.3.1.sip: SIP: SIP/2.0 180 Ringing 14:59:52.073381 IP 192.168.3.5.sip > 192.168.3.1.sip: SIP: SIP/2.0 183 Session Progress 15:00:02.027890 IP 192.168.3.5.sip > 192.168.3.1.sip: SIP: SIP/2.0 200 OK 15:00:02.028681 IP 192.168.3.1.sip > 192.168.3.5.sip: SIP: ACK sip:600@192.168.3.5:5060;transport=udp SIP/2.0 15:00:02.034884 IP 192.168.3.5.sip > 192.168.3.1.sip: SIP: BYE sip:423@192.168.3.1:5060 <http://sip:423@192.168.3.1:5060/> SIP/2.0 15:00:02.035367 IP 192.168.3.1.sip > 192.168.3.5.sip: SIP: SIP/2.0 200 OK
my supervisor tel me that when i change the file to .wav nrmally FS code and decode the file using unknown codec can you help me please sir ???
best regards
amani _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (2)
-
amani mansour
-
Michael Jerris