Re: [asterisk-users] Asterisk not using common codec between (SIP) endpoints
On Fri, Oct 4, 2019, at 1:45 AM, Andreas Wehrmann wrote: > > On 03/10/2019 16:24, Joshua C. Colp wrote: > > In PJSIP there is the PJSIP_MEDIA_OFFER dialplan function[1] but ultimately > > codec negotiation is not written or implemented in the way you need. There > > are some hints provided internally for outgoing legs but the result is > > still ultimately independent. That is: Each leg is negotiated from Asterisk > > to the endpoint, not endpoint to endpoint via Asterisk. This works for the > > vast majority of users as they have media flowing through Asterisk (by > > choice or via use of features) and are fine with transcoding (generally > > using codecs which aren't that costly or low channel count). > > > > Asterisk 16 has some of the foundational work to improve this through the > > implementation of streams but noone has worked on extending the codec > > negotiation support. > > > > [1] > > https://wiki.asterisk.org/wiki/display/AST/Asterisk+17+Function_PJSIP_MEDIA_OFFER > > > > Hey Joshua, > > do you think it might be possible to achieve this by writing a > supplement for the PJ part? PJSIP is only part of the equation, the information still has to transition across the core. > > What happens when the other side answers, but before the incoming call > is answered. > Is there a place in the code where, at that point, I have information > about both channels > and could theoretically influence the answer for the incoming call? Nope. That information is not currently exchanged or available, which is part of the problem. -- Joshua C. Colp Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk not using common codec between (SIP) endpoints
On 03/10/2019 16:24, Joshua C. Colp wrote: In PJSIP there is the PJSIP_MEDIA_OFFER dialplan function[1] but ultimately codec negotiation is not written or implemented in the way you need. There are some hints provided internally for outgoing legs but the result is still ultimately independent. That is: Each leg is negotiated from Asterisk to the endpoint, not endpoint to endpoint via Asterisk. This works for the vast majority of users as they have media flowing through Asterisk (by choice or via use of features) and are fine with transcoding (generally using codecs which aren't that costly or low channel count). Asterisk 16 has some of the foundational work to improve this through the implementation of streams but noone has worked on extending the codec negotiation support. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+17+Function_PJSIP_MEDIA_OFFER Hey Joshua, do you think it might be possible to achieve this by writing a supplement for the PJ part? What happens when the other side answers, but before the incoming call is answered. Is there a place in the code where, at that point, I have information about both channels and could theoretically influence the answer for the incoming call? All the best, Andy -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk not using common codec between (SIP) endpoints
On Thu, Oct 3, 2019, at 11:10 AM, Andreas Wehrmann wrote: > > On 03.10.19 15:08, Administrator TOOTAI wrote: > > > Before calling the gatreway add > > > > same = n,set(SIP_CODEC=alaw) > > > > [...] > > > > Hey there, > > that doesn't work as it seems to be implemented for chan_sip only; > I'm using chan_pjsip; sorry if I didn't explain myself properly. > > Anyway, in my case that would not really be an acceptable solution anyway, > because I need the called party to be able to pick from the range of > codecs presented to it > because the codec chosen by the destination might change (my example is > a simplified version). > > I don't think putting the burden of worrying about audio codecs on the > dialplan writer is a good idea, > since this should be dealt with automatically with respect to what is > configured and negotiated. > This is also because in the systems I have to work with, the 'engineers' > usual provide the configuration (endpoints, NAT config and the like) > while the technicians implement the dialplan (or the business logic so > to speak) according to customer needs. > They (the technicians) usually don't know (much) about codecs or how the > channels techs work exactly... In PJSIP there is the PJSIP_MEDIA_OFFER dialplan function[1] but ultimately codec negotiation is not written or implemented in the way you need. There are some hints provided internally for outgoing legs but the result is still ultimately independent. That is: Each leg is negotiated from Asterisk to the endpoint, not endpoint to endpoint via Asterisk. This works for the vast majority of users as they have media flowing through Asterisk (by choice or via use of features) and are fine with transcoding (generally using codecs which aren't that costly or low channel count). Asterisk 16 has some of the foundational work to improve this through the implementation of streams but noone has worked on extending the codec negotiation support. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+17+Function_PJSIP_MEDIA_OFFER -- Joshua C. Colp Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk not using common codec between (SIP) endpoints
On 03.10.19 15:08, Administrator TOOTAI wrote: Before calling the gatreway add same = n,set(SIP_CODEC=alaw) [...] Hey there, that doesn't work as it seems to be implemented for chan_sip only; I'm using chan_pjsip; sorry if I didn't explain myself properly. Anyway, in my case that would not really be an acceptable solution anyway, because I need the called party to be able to pick from the range of codecs presented to it because the codec chosen by the destination might change (my example is a simplified version). I don't think putting the burden of worrying about audio codecs on the dialplan writer is a good idea, since this should be dealt with automatically with respect to what is configured and negotiated. This is also because in the systems I have to work with, the 'engineers' usual provide the configuration (endpoints, NAT config and the like) while the technicians implement the dialplan (or the business logic so to speak) according to customer needs. They (the technicians) usually don't know (much) about codecs or how the channels techs work exactly... Thanks, Andy -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk not using common codec between (SIP) endpoints
Hi Le 03/10/2019 à 13:13, Andreas Wehrmann a écrit : [...] - Even if direct_media is disabled: Is there a way to make Asterisk always use a common codec between SIP endpoints, so it doesn't need to transcode? Before calling the gatreway add same = n,set(SIP_CODEC=alaw) [...] -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
