Re: [asterisk-users] Early audio SIP sequence order question
Hi Kevin I just found something interresting: http://www.faqs.org/rfc/rfc3960.txt 1. Unless a 180 (Ringing) response is received, never generate local ringing. 2. If a 180 (Ringing) has been received but there are no incoming media packets, generate local ringing. 3. If a 180 (Ringing) has been received and there are incoming media packets, play them and do not generate local ringing. So, yes, this most probably is an asterisk bug, if it's not a config issue I haven't figured out yet. Shall I submit a bug report? And indeed, if you dial more than one endpoint and more than one is sending early audio, which one do you forward? I think nobody tought about that issue. Well as long as one is being forwarded that would be ok for our case :-) Kind regards Benoit Panizzon -- I m p r o W a r e A G- __ Zurlindenstrasse 29 Tel +41 61 826 93 07 CH-4133 PrattelnFax +41 61 826 93 02 Schweiz Web http://www.imp.ch __ -- _ -- 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
Re: [asterisk-users] Early audio SIP sequence order question
On 02/11/2011 02:11 AM, Benoit Panizzon wrote: Hi Kevin I just found something interresting: http://www.faqs.org/rfc/rfc3960.txt 1. Unless a 180 (Ringing) response is received, never generate local ringing. 2. If a 180 (Ringing) has been received but there are no incoming media packets, generate local ringing. 3. If a 180 (Ringing) has been received and there are incoming media packets, play them and do not generate local ringing. So, yes, this most probably is an asterisk bug, if it's not a config issue I haven't figured out yet. Shall I submit a bug report? It is possible that the 'progressinband' configuration option is related to this, but I'm not sure that it is (since I believe it only reacts to 183 messages, not 180). Given that, it would be best to open a bug report to get it in the queue for someone to look at. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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
Re: [asterisk-users] Early audio SIP sequence order question
Go here http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial and use proper parameters to dial command to pass early media. -Original Message- From: Benoit Panizzon Sent: Thursday, February 10, 2011 4:08 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Early audio SIP sequence order question Hello We have quite some problems with early audio with our asterisk 1.6.2.15 What we observe is: Asterisk - Carrier PBX Asterisk:Invite(+sdp) = Carrier Carrier starts to send RTP Audio (ignored by Asterisk) Asterisk = Carrier:100 Trying Asterisk = Carrier:180 Ringing Asterisk signals Ringing to the caller which in turn generated the ringing tone (still ignoring the early audio sent by the carrier). Asterisk = Carrier:200 OK(+sdp) Asterisk:ACK = Carrier Asterisk starts to send RTP Audio to Carrier Only now Asterisk starts playing Audio to the caller. This causes quite troubles, as the price of a value added number is announced in early audio in switzerland, giving the caller a chance to hang up before the call is established. But the caller connected to asterisk does not hear that early audio announcement. Is this an asterisk bug, or should the carrier have signaled 183 Session Progress instead of 180 Ringing? Kind regards Benoit Panizzon -- I m p r o W a r e A G- __ Zurlindenstrasse 29 Tel +41 61 826 93 07 CH-4133 PrattelnFax +41 61 826 93 02 Schweiz Web http://www.imp.ch __ -- _ -- 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
Re: [asterisk-users] Early audio SIP sequence order question
Hi Faisal http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial Thank you, but I seem to miss the option which tells asterisk to pass audio even if no 183 or 200 is received. No, we don't set the 'r' Flag while dialing out. So, my question ist sill the same. Sould asterisk pass audio of it didn't yet receive a 183 or 200 message, or is the carrier doing wrong in sending early audio without 183? Kind regards Benoit Panizzon -- I m p r o W a r e A G- __ Zurlindenstrasse 29 Tel +41 61 826 93 07 CH-4133 PrattelnFax +41 61 826 93 02 Schweiz Web http://www.imp.ch __ -- _ -- 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
Re: [asterisk-users] Early audio SIP sequence order question
On 02/10/2011 06:15 AM, Benoit Panizzon wrote: Hi Faisal http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial Thank you, but I seem to miss the option which tells asterisk to pass audio even if no 183 or 200 is received. No, we don't set the 'r' Flag while dialing out. So, my question ist sill the same. Sould asterisk pass audio of it didn't yet receive a 183 or 200 message, or is the carrier doing wrong in sending early audio without 183? This does indeed sound like an Asterisk bug; Asterisk should be ready and willing to accept audio from the called SIP endpoint as soon as the INVITE is sent out with an SDP offer to receive audio. Now the real issue here may be the Dial() application not forwarding that audio to the caller, rather than Asterisk not 'accepting' the audio and turning it into internal media frames. The net result for you is the same, but the source of the problem is quite different. This can of course cause complications if Dial() is used to dial multiple endpoints... because then there could be multiple audio streams received from them as the call proceeds towards one of them answering. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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
