Hello Phillip! Thank you.
However, do you what Asterisk does if there is no rtpmap that describes the format profile in the media description? The SIP packets states: audio 25184 RTP/AVP 18 98 96 97 101 13 However, there is no rtpmap for format 18. Asterisk does not seem to be associating 18 with the G729 codec, since the calls fails at pickup. Thank you, Elliot On Thu, Jul 2, 2009 at 11:15 AM, Philipp Kempgen<philipp.kemp...@amooma.de> wrote: > Elliot Murdock schrieb: >> Thank you for that piece of information. Which RFC does it state that >> the audio name is "G729"? > > http://tools.ietf.org/html/rfc3555#section-4.1.9 > >> On Thu, Jul 2, 2009 at 12:16 AM, Kevin P. Fleming<kpflem...@digium.com> >> wrote: >>> Elliot Murdock wrote: >>>> 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. > > http://tools.ietf.org/html/rfc3555#section-3 > ---cut--- > Note that the payload format (encoding) names defined in the RTP > Profile are commonly shown in upper case. MIME subtypes are commonly > shown in lower case. These names are case-insensitive in both > places. > ---cut--- > > Sounds like it should really be case-insensitive but I might easily > be mistaken. Didn't dig too deep into RTP/SDP specifications. > > > Philipp Kempgen > -- > AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de > Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 > Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de > Videos of the AMOOCON VoIP conference 2009 -> http://www.amoocon.de > -- > > _______________________________________________ > -- 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
