Hello, It is my understanding that on the softphone side, asterisk is only responsible for establishing the session between two phones. If this is the case, does it matter what type of audio codecs the two phones are using? And if it does matter, are there any codecs that cause problems with asterisk bridging two SIP connections? Thanks for your helpful input, Daniel
On Thu, 2003-07-03 at 15:01, Daniel Flickinger wrote:> Hello, > > It is my understanding that on the softphone side, asterisk is only > responsible for establishing the session between two phones. If this is the > case, does it matter what type of audio codecs the two phones are using? And > if it does matter, are there any codecs that cause problems with asterisk > bridging two SIP connections? Thanks for your helpful input,This depends. If a SIP phone must use the IVR feature of asterisk to get routed to another SIP phone, then codecs matter. If asterisk is listening on the line to hear when you issue commands via DTMF for it to do something like transfer, then yes it matters. Also some SIP phones don't handle reinvite properly, and then asterisk is stuck redirecting the audio from place to place for you, here it doesn't matter unless on of the above comments apply. -- Steven Critchfield <critch@basesys.com>
Thank you for your help Steven.> Message: 8 > Subject: Re: [Asterisk-Users] Drops due to codecs? > From: Steven Critchfield <critch@basesys.com> > To: asterisk-users@lists.digium.com > Date: 03 Jul 2003 15:26:45 -0500> Reply-To: asterisk-users@lists.digium.com> On Thu, 2003-07-03 at 15:01, Daniel Flickinger wrote: > > Hello, > > > It is my understanding that on the softphone side, asterisk is only > > responsible for establishing the session between two phones. If this isthe case, does it matter what type of audio codecs the two phones are using? And> if it does matter, are there any codecs that cause problems with asterisk > bridging two SIP connections? Thanks for your helpful input,> This depends. If a SIP phone must use the IVR feature of asterisk to get > routed to another SIP phone, then codecs matter. If asterisk is > listening on the line to hear when you issue commands via DTMF for it to > do something like transfer, then yes it matters. Also some SIP phones > don't handle reinvite properly, and then asterisk is stuck redirecting > the audio from place to place for you, here it doesn't matter unless on > of the above comments apply. > -- Steven Critchfield <critch@basesys.com>