Ray Jackson
2006-Nov-19 03:59 UTC
[asterisk-users] G723 pass-through and codec negotiation
All, Our users have a softphone client that supports the G723 Codec which we want to use for bandwidth reasons, however we do not wish (or have the funds) to license the codec on our Asterisk servers. We have G723 pass-through working between the clients just fine, however calls fail when terminating with Asterisk itself (i.e. Voicemail) or out to the PSTN due to transcoding issues. If it possible to build the config into our Asterisk servers so that calls between the softphones defaults to G723 pass-through, whilst all other calls (PSTN, Voicemail etc.) default to GSM as their preferred codec? Is there a way of getting Asterisk to be smart with Codec negotitation and figure out which codec the other end of the call is capable of before negotitating back to the Softphone with the selected codec? I assume you would have to do something in the dial plan? I saw the SIP_CODEC variable, but couldn't make it work. Any advice would be very welcome! Cheers, Ray ----------------------------------------------------------------------------------------------- This message and any attachments contain privileged and confidential information. If you are not the intended recipient of this message, you are hereby notified that any use, dissemination, distribution or reproduction of this message is prohibited. If you have received this message in error please notify the sender immediately via email and then destroy this message and any attachments.
Andrew Joakimsen
2006-Nov-19 08:47 UTC
[asterisk-users] G723 pass-through and codec negotiation
What happens if in your sip.conf you set disallow=all allow=g723,gsm And then allow both codec in the phone? On 11/19/06, Ray Jackson <ray@jacksonz.net> wrote:> > All, > > Our users have a softphone client that supports the G723 Codec which we > want to use for bandwidth reasons, however we do not wish (or have the > funds) to license the codec on our Asterisk servers. We have G723 > pass-through working between the clients just fine, however calls fail > when terminating with Asterisk itself (i.e. Voicemail) or out to the > PSTN due to transcoding issues. > > If it possible to build the config into our Asterisk servers so that > calls between the softphones defaults to G723 pass-through, whilst all > other calls (PSTN, Voicemail etc.) default to GSM as their preferred > codec? Is there a way of getting Asterisk to be smart with Codec > negotitation and figure out which codec the other end of the call is > capable of before negotitating back to the Softphone with the selected > codec? I assume you would have to do something in the dial plan? I saw > the SIP_CODEC variable, but couldn't make it work. > > Any advice would be very welcome! > > Cheers, > Ray > > > ----------------------------------------------------------------------------------------------- > This message and any attachments contain privileged and confidential > information. > If you are not the intended recipient of this message, you are hereby > notified that any use, dissemination, distribution or reproduction of this > message is prohibited. If you have received this message in error please > notify the sender immediately via email and then destroy this message and > any attachments. > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061119/db8f1bfd/attachment.htm