We are getting ready to install a client that uses g729 when talking to their SIP provider to minimize bandwidth usage. We are going to want to be able to record the calls using AMI monitor actions into wav sound files. All the phones are soft phone running on Windows XP systems. Questions I have are what would the best codec be to have the soft phone use since, as I understand it, in order to mix the audio something will need to be transcoded. Can a two CPU quad core xeon 2GHz system handle the transcoding load or would if be better to have a daughter card handle the transcoding. -- Jim Dickenson mailto:dickenson at cfmc.com CfMC http://www.cfmc.com/
There will be up to 150 phones so there will be 300 channels when they are all on the phone at one time. I will be using a current 1.4 version. -- Jim Dickenson mailto:dickenson at cfmc.com CfMC http://www.cfmc.com/ On Mar 22, 2010, at 5:05 PM, Rafael Prado Rocchi wrote:> How many simultaneous channels? > > Rafael Prado > > >> -----Original Message----- >> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users- >> bounces at lists.digium.com] On Behalf Of Jim Dickenson >> Sent: segunda-feira, 22 de mar?o de 2010 2:33 >> To: Asterisk User MailList >> Subject: [asterisk-users] Transcoding question >> >> We are getting ready to install a client that uses g729 when talking to >> their SIP provider to minimize bandwidth usage. We are going to want to >> be able to record the calls using AMI monitor actions into wav sound >> files. All the phones are soft phone running on Windows XP systems. >> >> Questions I have are what would the best codec be to have the soft >> phone use since, as I understand it, in order to mix the audio >> something will need to be transcoded. Can a two CPU quad core xeon 2GHz >> system handle the transcoding load or would if be better to have a >> daughter card handle the transcoding. >> >> >> -- >> Jim Dickenson >> mailto:dickenson at cfmc.com >> >> CfMC >> http://www.cfmc.com/ >> >> >> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
How many simultaneous channels? Rafael Prado> -----Original Message----- > From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users- > bounces at lists.digium.com] On Behalf Of Jim Dickenson > Sent: segunda-feira, 22 de mar?o de 2010 2:33 > To: Asterisk User MailList > Subject: [asterisk-users] Transcoding question > > We are getting ready to install a client that uses g729 when talking to > their SIP provider to minimize bandwidth usage. We are going to want to > be able to record the calls using AMI monitor actions into wav sound > files. All the phones are soft phone running on Windows XP systems. > > Questions I have are what would the best codec be to have the soft > phone use since, as I understand it, in order to mix the audio > something will need to be transcoded. Can a two CPU quad core xeon 2GHz > system handle the transcoding load or would if be better to have a > daughter card handle the transcoding. > > > -- > Jim Dickenson > mailto:dickenson at cfmc.com > > CfMC > http://www.cfmc.com/ > > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 5192 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20100322/9bf2b9df/attachment.bin