Re: [asterisk-users] Inquiry:Asterisk different codec schemes?
4 jan 2010 kl. 14.46 skrev Kevin P. Fleming: hadi motamedi wrote: Sorry . I didn't get the point clearly . In the SIP Invite message , it says my audio endpoint is IP x.x.x.x port x, and I can use codecs A,B,C. The remote endpoint responds with a 200 OK, saying my audio stream is at IP y.y.y.y port y, and I choose codec B. Can you please do me favor and let me know if my understanding is right or not ? Thank you No, you are not understanding the SDP offer/answer model properly. If one endpoint offers codecs A, B and C in its SDP, it is willing to *receive* media in those formats. The receiver of that offer can choose to send media to the offerer in any of those formats, at any time. If the answering endpoint includes only codec B in its SDP, then it is willing to *receive* only codec B. In that scenario, it is possible for media to flow from endpoint 1 to endpoint 2 using codec B, and from endpoint 2 to endpoint 1 using codec A (or C), but this will not happen if Asterisk is an endpoint in this scenario. When Asterisk receives a media frame, if the format of that frame is not the format that it is currently sending to the other endpoint, it will switch to that format automatically. If it cannot do so because the other endpoint did not offer to receive that format, then the call's audio will probably fail. This is the reason why I responded before that Asterisk does not support asymmetric formats in a media session. In reality, it is extremely uncommon for a SIP endpoint to want to send media in a format that it is not also willing to receive; in fact, I can't say I've ever seen this situation arise in any testing I've done or in any issues reported in our issue tracker. But it's fairly common to have asymmetric media in the call. If the caller offers A, B and C and the callee responds with B, the caller sends B but the callee might send A. /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Asterisk different codec schemes?
On Tue, Jan 5, 2010 at 8:59 AM, Olle E. Johansson o...@edvina.net wrote: 4 jan 2010 kl. 14.46 skrev Kevin P. Fleming: hadi motamedi wrote: Sorry . I didn't get the point clearly . In the SIP Invite message , it says my audio endpoint is IP x.x.x.x port x, and I can use codecs A,B,C. The remote endpoint responds with a 200 OK, saying my audio stream is at IP y.y.y.y port y, and I choose codec B. Can you please do me favor and let me know if my understanding is right or not ? Thank you No, you are not understanding the SDP offer/answer model properly. If one endpoint offers codecs A, B and C in its SDP, it is willing to *receive* media in those formats. The receiver of that offer can choose to send media to the offerer in any of those formats, at any time. If the answering endpoint includes only codec B in its SDP, then it is willing to *receive* only codec B. In that scenario, it is possible for media to flow from endpoint 1 to endpoint 2 using codec B, and from endpoint 2 to endpoint 1 using codec A (or C), but this will not happen if Asterisk is an endpoint in this scenario. When Asterisk receives a media frame, if the format of that frame is not the format that it is currently sending to the other endpoint, it will switch to that format automatically. If it cannot do so because the other endpoint did not offer to receive that format, then the call's audio will probably fail. This is the reason why I responded before that Asterisk does not support asymmetric formats in a media session. In reality, it is extremely uncommon for a SIP endpoint to want to send media in a format that it is not also willing to receive; in fact, I can't say I've ever seen this situation arise in any testing I've done or in any issues reported in our issue tracker. But it's fairly common to have asymmetric media in the call. If the caller offers A, B and C and the callee responds with B, the caller sends B but the callee might send A. /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Sorry . You mean we can have asymmetric codecs in Asterisk ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Asterisk different codec schemes?
Olle E. Johansson wrote: But it's fairly common to have asymmetric media in the call. If the caller offers A, B and C and the callee responds with B, the caller sends B but the callee might send A. Only for non-Asterisk endpoints, since Asterisk will never do this. Is this really that common? I'd be surprised if an endpoint would want to consume a G.729 encoder (for example) without a corresponding decoder on the receive path... doing that would make managing DSP resources in the endpoint much more complicated. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Asterisk different codec schemes?
5 jan 2010 kl. 10.08 skrev hadi motamedi: On Tue, Jan 5, 2010 at 8:59 AM, Olle E. Johansson o...@edvina.net wrote: 4 jan 2010 kl. 14.46 skrev Kevin P. Fleming: hadi motamedi wrote: Sorry . I didn't get the point clearly . In the SIP Invite message , it says my audio endpoint is IP x.x.x.x port x, and I can use codecs A,B,C. The remote endpoint responds with a 200 OK, saying my audio stream is at IP y.y.y.y port y, and I choose codec B. Can you please do me favor and let me know if my understanding is right or not ? Thank you No, you are not understanding the SDP offer/answer model properly. If one endpoint offers codecs A, B and C in its SDP, it is willing to *receive* media in those formats. The receiver of that offer can choose to send media to the offerer in any of those formats, at any time. If the answering endpoint includes only codec B in its SDP, then it is willing to *receive* only codec B. In that scenario, it is possible for media to flow from endpoint 1 to endpoint 2 using codec B, and from endpoint 2 to endpoint 1 using codec A (or C), but this will not happen if Asterisk is an endpoint in this scenario. When Asterisk receives a media frame, if the format of that frame is not the format that it is currently sending to the other endpoint, it will switch to that format automatically. If it cannot do so because the other endpoint did not offer to receive that format, then the call's audio will probably fail. This is the reason why I responded before that Asterisk does not support asymmetric formats in a media session. In reality, it is extremely uncommon for a SIP endpoint to want to send media in a format that it is not also willing to receive; in fact, I can't say I've ever seen this situation arise in any testing I've done or in any issues reported in our issue tracker. But it's fairly common to have asymmetric media in the call. If the caller offers A, B and C and the callee responds with B, the caller sends B but the callee might send A. /O Sorry . You mean we can have asymmetric codecs in Asterisk ? As Kevin stated, for Asterisk, the server switches to the format we receive, so no. I just pointed out that it happens quite often that a call is asymmetric, and you will see Asterisk trying to follow the other side. /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Asterisk different codec schemes?
hadi motamedi wrote: Sorry . I didn't get the point clearly . In the SIP Invite message , it says my audio endpoint is IP x.x.x.x port x, and I can use codecs A,B,C. The remote endpoint responds with a 200 OK, saying my audio stream is at IP y.y.y.y port y, and I choose codec B. Can you please do me favor and let me know if my understanding is right or not ? Thank you No, you are not understanding the SDP offer/answer model properly. If one endpoint offers codecs A, B and C in its SDP, it is willing to *receive* media in those formats. The receiver of that offer can choose to send media to the offerer in any of those formats, at any time. If the answering endpoint includes only codec B in its SDP, then it is willing to *receive* only codec B. In that scenario, it is possible for media to flow from endpoint 1 to endpoint 2 using codec B, and from endpoint 2 to endpoint 1 using codec A (or C), but this will not happen if Asterisk is an endpoint in this scenario. When Asterisk receives a media frame, if the format of that frame is not the format that it is currently sending to the other endpoint, it will switch to that format automatically. If it cannot do so because the other endpoint did not offer to receive that format, then the call's audio will probably fail. This is the reason why I responded before that Asterisk does not support asymmetric formats in a media session. In reality, it is extremely uncommon for a SIP endpoint to want to send media in a format that it is not also willing to receive; in fact, I can't say I've ever seen this situation arise in any testing I've done or in any issues reported in our issue tracker. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Asterisk different codec schemes?
On Mon, Jan 4, 2010 at 1:46 PM, Kevin P. Fleming kpflem...@digium.comwrote: hadi motamedi wrote: Sorry . I didn't get the point clearly . In the SIP Invite message , it says my audio endpoint is IP x.x.x.x port x, and I can use codecs A,B,C. The remote endpoint responds with a 200 OK, saying my audio stream is at IP y.y.y.y port y, and I choose codec B. Can you please do me favor and let me know if my understanding is right or not ? Thank you No, you are not understanding the SDP offer/answer model properly. If one endpoint offers codecs A, B and C in its SDP, it is willing to *receive* media in those formats. The receiver of that offer can choose to send media to the offerer in any of those formats, at any time. If the answering endpoint includes only codec B in its SDP, then it is willing to *receive* only codec B. In that scenario, it is possible for media to flow from endpoint 1 to endpoint 2 using codec B, and from endpoint 2 to endpoint 1 using codec A (or C), but this will not happen if Asterisk is an endpoint in this scenario. When Asterisk receives a media frame, if the format of that frame is not the format that it is currently sending to the other endpoint, it will switch to that format automatically. If it cannot do so because the other endpoint did not offer to receive that format, then the call's audio will probably fail. This is the reason why I responded before that Asterisk does not support asymmetric formats in a media session. In reality, it is extremely uncommon for a SIP endpoint to want to send media in a format that it is not also willing to receive; in fact, I can't say I've ever seen this situation arise in any testing I've done or in any issues reported in our issue tracker. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thank you very much for correcting me . ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Asterisk different codec schemes?
On Thu, Dec 31, 2009 at 12:10 PM, Kevin P. Fleming kpflem...@digium.comwrote: hadi motamedi wrote: Can you please let me know if we can have different codec schemes for audio codec in audio codec out ? I mean , in one application , we can have our audio codec input set to G.711 a-law and our audio codec output set to G.711 u-law . I am facing with an application that calls for such a settings . Asterisk does not support asymmetric codec configurations. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Sorry . I didn't get the point clearly . In the SIP Invite message , it says my audio endpoint is IP x.x.x.x port x, and I can use codecs A,B,C. The remote endpoint responds with a 200 OK, saying my audio stream is at IP y.y.y.y port y, and I choose codec B. Can you please do me favor and let me know if my understanding is right or not ? Thank you ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Asterisk different codec schemes?
hadi motamedi wrote: Can you please let me know if we can have different codec schemes for audio codec in audio codec out ? I mean , in one application , we can have our audio codec input set to G.711 a-law and our audio codec output set to G.711 u-law . I am facing with an application that calls for such a settings . Asterisk does not support asymmetric codec configurations. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Inquiry:Asterisk different codec schemes?
Dear All Can you please let me know if we can have different codec schemes for audio codec in audio codec out ? I mean , in one application , we can have our audio codec input set to G.711 a-law and our audio codec output set to G.711 u-law . I am facing with an application that calls for such a settings . Thank you ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
