Hey Flavio! Hope all is well with you (aside from this little SDP issue). Looking forward to seeing you guys at ClueCon 2013. :) Are you able to reproduce this issue on latest git HEAD? If so, could you open a ticket at jira.freeswitch.org please? That will be the quickest way to get it resolved. Thanks! -Michael On Mon, Feb 25, 2013 at 6:08 AM, Flavio Goncalves <flavio@sippulse.com>wrote:
Hi,
I have searched the list, but I couldn't find a final answer for this case. I'm receiving calls from a provider using G729 annex B.
INVITE a=rtpmap:18 G729/8000. a=fmtp:18 annexb=yes. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=rtpmap:19 CN/8000.
and FS is answering
200OK m=audio 31388 RTP/AVP 18 101 19. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=yes. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=rtpmap:19 CN/8000. a=ptime:20.
If freeswitch supports only G729A, shouldn't FS refuse the call with 488 or suppress the line annexb=yes?
The audio quality is really bad when I receive calls in this way. After removing the annexb on the gateway the audio is fine, but unfortunately I can't remove from it from all VoIP providers terminating calls. For outbound calls I simply removed the annexb, but for inbound there is no way to do it.
Flavio
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