covici at ccs.covici.com
2012-Jan-01 22:17 UTC
[asterisk-users] asterisk 1.8 codec negotiation
Hi. I am using asterisk 1.8 and everything was working fine when I was at svn 342661. I then upgraded to vrsion 349339 and discovered the following problem -- one of the end points is a freeswitch box which offers a number of codecs, including PCMU. However, when I tried to make a call I got a 488 response and a message "multiple audio streams not supported" in the log. Is this by design? I found an issue 18859, but that referenced where the end point offered both regular rtp and srtp. But it seems to me if an endpoint offers various codecs, that asterisk could only complain if none of them match one that asterisk likes. If I only offer one codec, it works, but that seems an unnecessary restriction to me. Any assistance on this would be appreciated. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com
José Pablo Méndez Soto
2012-Jan-02 05:15 UTC
[asterisk-users] asterisk 1.8 codec negotiation
Can you show us how the previous INVITE Looked like vs the current one? *Jos? Pablo M?ndez ********* On Sun, Jan 1, 2012 at 4:17 PM, <covici at ccs.covici.com> wrote:> Hi. I am using asterisk 1.8 and everything was working fine when I was > at svn 342661. I then upgraded to vrsion 349339 and discovered the > following problem -- one of the end points is a freeswitch box which > offers a number of codecs, including PCMU. However, when I tried to > make a call I got a 488 response and a message "multiple audio streams > not supported" in the log. > > Is this by design? I found an issue 18859, but that referenced where > the end point offered both regular rtp and srtp. But it seems to me if > an endpoint offers various codecs, that asterisk could only complain if > none of them match one that asterisk likes. > > If I only offer one codec, it works, but that seems an unnecessary > restriction to me. > > Any assistance on this would be appreciated. > > -- > Your life is like a penny. You're going to lose it. The question is: > How do > you spend it? > > John Covici > covici at ccs.covici.com > > -- > _____________________________________________________________________ > -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120101/c2357011/attachment.htm>
On 01/01/2012 04:17 PM, covici at ccs.covici.com wrote:> Hi. I am using asterisk 1.8 and everything was working fine when I was > at svn 342661. I then upgraded to vrsion 349339 and discovered the > following problem -- one of the end points is a freeswitch box which > offers a number of codecs, including PCMU. However, when I tried to > make a call I got a 488 response and a message "multiple audio streams > not supported" in the log."multiple audio streams" != "multiple audio codecs". For some reason Asterisk is receiving an INVITE with an offer for more than one audio stream (m=audio), and that is not supported. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kfleming at digium.com | SIP: kpfleming at digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com & www.asterisk.org