Hello! I have a sip device that is sending in the SDP: rtpmap:98 g729a It does not seem like Asterisk is negotiating the codec properly, because while the call rings, the rtp lines fail. However, on other sip devices that have "rtpmap:18 g729" in their SDP, things work fine with Digium's commercial g729 license. How do I get "98 g729a" recognized by Asterisk? Thanks, Elliot
Elliot Murdock wrote:> Hello! > > I have a sip device that is sending in the SDP: > > rtpmap:98 g729a > > It does not seem like Asterisk is negotiating the codec properly, > because while the call rings, the rtp lines fail. However, on other > sip devices that have "rtpmap:18 g729" in their SDP, things work fine > with Digium's commercial g729 license. > > How do I get "98 g729a" recognized by Asterisk?You don't. That's not a standards-compliant way of reporting G.729A in SDP. The RFC says it should be 'G729', but Asterisk also accepts 'G.729' and 'G729A'. It does not accept any lowercase form of the codec name. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpfleming at digium.com Check us out at www.digium.com & www.asterisk.org
Hello, Thank you clarifying that. However, if that is the case, why is Asterisk sending back PCMU packets (instead of G729), which the device is not enabled for and subsequently, fails the call? Could the mapping be disabled or not properly mapping to the G729 driver in a certain versions of Asterisk? Thanks, Elliot On Thu, Jul 2, 2009 at 3:25 PM, Kevin P. Fleming<kpfleming at digium.com> wrote:> Elliot Murdock wrote: >> Hello! >> >> I noticed that the SIP packet contains this line: >> >> m=audio 60000 RTP/AVP 18 98 96 97 101 13 >> >> However, there is no rtpmap that describes 18. ?Media format 18 >> Apparently refers to G729, but there is no rtpmap in the SDP for it. >> Since G729 is a registered and known format is there any way for >> Asterisk to negotiate it within an explicit rtpmap? > > Yes, that is already supported. Asterisk does not require rtpmap entries > for well-known (RFC specified) codec mappings. > > -- > Kevin P. Fleming > Digium, Inc. | Director of Software Technologies > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > skype: kpfleming | jabber: kpfleming at 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 >
Hello! Which RFC specifies the corresponding number of the formats? Where in the Asterisk source code does it state the SDP formats? Does Asterisk follow the formats of IANA? (http://www.iana.org/assignments/rtp-parameters) Thank you, Elliot On Thu, Jul 2, 2009 at 3:44 PM, Elliot Murdock<murdocke at gmail.com> wrote:> Hello, > > Thank you clarifying that. > > However, if that is the case, why is Asterisk sending back PCMU > packets (instead of G729), which the device is not enabled for and > subsequently, fails the call? > > Could the mapping be disabled or not properly mapping to the G729 > driver in a certain versions of Asterisk? > > Thanks, > Elliot > > On Thu, Jul 2, 2009 at 3:25 PM, Kevin P. Fleming<kpfleming at digium.com> wrote: >> Elliot Murdock wrote: >>> Hello! >>> >>> I noticed that the SIP packet contains this line: >>> >>> m=audio 60000 RTP/AVP 18 98 96 97 101 13 >>> >>> However, there is no rtpmap that describes 18. ?Media format 18 >>> Apparently refers to G729, but there is no rtpmap in the SDP for it. >>> Since G729 is a registered and known format is there any way for >>> Asterisk to negotiate it within an explicit rtpmap? >> >> Yes, that is already supported. Asterisk does not require rtpmap entries >> for well-known (RFC specified) codec mappings. >> >> -- >> Kevin P. Fleming >> Digium, Inc. | Director of Software Technologies >> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA >> skype: kpfleming | jabber: kpfleming at 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 >> >
On Thu, 2 Jul 2009, Elliot Murdock wrote:> Hello! > > Which RFC specifies the corresponding number of the formats? > > Where in the Asterisk source code does it state the SDP formats? > > Does Asterisk follow the formats of IANA? > (http://www.iana.org/assignments/rtp-parameters) > > Thank you, > ElliotPerhaps this is falling back too far, but do you have G.729 licenses for your asterisk server? j> > > On Thu, Jul 2, 2009 at 3:44 PM, Elliot Murdock<murdocke at gmail.com> wrote: >> Hello, >> >> Thank you clarifying that. >> >> However, if that is the case, why is Asterisk sending back PCMU >> packets (instead of G729), which the device is not enabled for and >> subsequently, fails the call? >> >> Could the mapping be disabled or not properly mapping to the G729 >> driver in a certain versions of Asterisk? >> >> Thanks, >> Elliot >> >> On Thu, Jul 2, 2009 at 3:25 PM, Kevin P. Fleming<kpfleming at digium.com> wrote: >>> Elliot Murdock wrote: >>>> Hello! >>>> >>>> I noticed that the SIP packet contains this line: >>>> >>>> m=audio 60000 RTP/AVP 18 98 96 97 101 13 >>>> >>>> However, there is no rtpmap that describes 18. ?Media format 18 >>>> Apparently refers to G729, but there is no rtpmap in the SDP for it. >>>> Since G729 is a registered and known format is there any way for >>>> Asterisk to negotiate it within an explicit rtpmap? >>> >>> Yes, that is already supported. Asterisk does not require rtpmap entries >>> for well-known (RFC specified) codec mappings. >>> >>> -- >>> Kevin P. Fleming >>> Digium, Inc. | Director of Software Technologies >>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA >>> skype: kpfleming | jabber: kpfleming at 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 >>> >> > > _______________________________________________ > -- 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 >