.wav/.mp3 recording problems - lag
Hello, I recently set up a FS system for one call-center. One of the request was to enable call recording for inbound calls (it's just an inbound callcenter (20 lines) no outbound calls). At first i went with .mp3 enconding but i soon found out that in a "longer" conversation 8-9 minutes the two channels would get very much lagged.T The caller's, leg A, questions would be far ahead of the callees', leg B, answers. I though that this was caused by the .mp3 encoding so i switched back to plain old .wav. The problem is that the lag is still there and comes to "life" totally random. Has anyone ever encountered this before ? Thanks, Bogdan
mp3 encoding requires some use of resources, as does recording (in any format). Have you considered using a separate system to record the calls and dedicating resources to allow this? On Wed, Jul 14, 2010 at 7:50 AM, Bogdan Patrascoiu < bogdan.patrascoiu@sinergetic.ro> wrote:
Hello,
I recently set up a FS system for one call-center. One of the request was to enable call recording for inbound calls (it's just an inbound callcenter (20 lines) no outbound calls). At first i went with .mp3 enconding but i soon found out that in a "longer" conversation 8-9 minutes the two channels would get very much lagged.T The caller's, leg A, questions would be far ahead of the callees', leg B, answers. I though that this was caused by the .mp3 encoding so i switched back to plain old .wav. The problem is that the lag is still there and comes to "life" totally random.
Has anyone ever encountered this before ?
Thanks,
Bogdan
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A couple of questions and some suggestions. I assume that you are connecting an incoming call to a local operator. Which means that you actually are using two channels. Since you are doing ACD work, you probably want to do call transfers from one operator to another. You may also want to allow call monitoring by a supervisor. This means that you may at times have more than two channels active on the call. 1. How are you doing the recording? 2. I may be wrong, but I think that the proper way to connect two channels together is to bridge them. This is a form of conferencing. If you are not doing the recording from the conference, why not change this. Then everything in the conference would be recorded. 3. Consider recording the information in WAV format. One note on some Wiki page comments that disk is now cheap and fairly plentiful. Why waste FreeSWITCH real-time resources to create an MP3 file, record it in WAV. 4. If you want an mp3 file, find a converter from WAV to MP3 or some other format. Run this after the call/conference is over. Note: MP3 does have some license considerations so if compression is the main consideration, then why not use some newer format that does not have license considerations and may actually give better compression. Note: For some other reasons, we are going to write an Erlang program that connects via mod_erlang_event and will be able to determine the end of a conference and do any necessary post processing. If we were to need to compress the recording, this is where we would do it. You could even put part the Erlang program on a separate system and put the compressed recording on the other system. Then once the file has been copied/compressed, it could be deleted from the original system. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/wav-mp3-recording-problems-lag... Sent from the freeswitch-users mailing list archive at Nabble.com.
If you can find a way to reproduce it manually we can have a look at it. I assume this is on the latest GIT version? On Wed, Jul 14, 2010 at 6:50 AM, Bogdan Patrascoiu < bogdan.patrascoiu@sinergetic.ro> wrote:
Hello,
I recently set up a FS system for one call-center. One of the request was to enable call recording for inbound calls (it's just an inbound callcenter (20 lines) no outbound calls). At first i went with .mp3 enconding but i soon found out that in a "longer" conversation 8-9 minutes the two channels would get very much lagged.T The caller's, leg A, questions would be far ahead of the callees', leg B, answers. I though that this was caused by the .mp3 encoding so i switched back to plain old .wav. The problem is that the lag is still there and comes to "life" totally random.
Has anyone ever encountered this before ?
Thanks,
Bogdan
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
Hello, My dial plan looks something like this : <extension name="provider" > <condition field="destination_number" expression="^(.+),(.+),(.{10})$"> <action application="set" data="record_file=$${base_dir}/recordings/$1_$2_${strftime(%Y-%m-%d-%H-%M-%S)}_$3_${caller_id_number}.wav"/> <action application="set" data="tariff=0.02"/> <action application="set" data="id_proiect=$1"/> <action application="set" data="id_operator=$2"/> <action application="set" data="real_destination=$1,$2,$3"/> <action application="set" data="transfer_after_bridge=*3472"/> <action application="bind_meta_app" data="0 a s transfer::*3472"/> <action application="record_session" data="${record_file}"/> <action application="bridge" data="sofia/gateway/53625621${caller_id_number}/$3"/> </condition> </extension> ~ After a lot of digging i found this : http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg10499.h... So the answer seems to be this : <action application="set" data="enable_file_write_buffering=false"/> What do you think? I implemented this and so far i haven't got any new complains, but as the happening was totally random i am not sure of this solution's certainty. Thanks, Bogdan On Wed, Jul 14, 2010 at 6:36 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
If you can find a way to reproduce it manually we can have a look at it. I assume this is on the latest GIT version?
On Wed, Jul 14, 2010 at 6:50 AM, Bogdan Patrascoiu <bogdan.patrascoiu@sinergetic.ro> wrote:
Hello,
I recently set up a FS system for one call-center. One of the request was to enable call recording for inbound calls (it's just an inbound callcenter (20 lines) no outbound calls). At first i went with .mp3 enconding but i soon found out that in a "longer" conversation 8-9 minutes the two channels would get very much lagged.T The caller's, leg A, questions would be far ahead of the callees', leg B, answers. I though that this was caused by the .mp3 encoding so i switched back to plain old .wav. The problem is that the lag is still there and comes to "life" totally random.
Has anyone ever encountered this before ?
Thanks,
Bogdan
_______________________________________________ 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
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
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participants (4)
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Anthony Minessale
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Bogdan Patrascoiu
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David Swardstrom
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Tony Graziano