<div dir="ltr">Hi Brian, <div><br></div><div style>This setting was already set. For outbound calls it works just fine. For inbound calls, I can't disable the annexb. I need to reply (200 Ok) with annexb=no.</div><div style>
<br></div><div style>Best regards, </div><div class="gmail_extra"><br clear="all"><div><div dir="ltr">Flavio E. Goncalves<br>CTO - SipPulse Routing and Billing Solutions for SIP<br></div></div><br><div class="gmail_quote">
2013/2/25 Brian West <span dir="ltr"><<a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Try setting this variable:<br>
<br>
sip_append_audio_sdp=a=fmtp:18 annexb=no<br>
<br>
Can you verify that your provider is actually sending CN on 19 and NOT 13. There are some Sonus switches that say 19 but mean 13 and send on 13. 19=13 in hex.<br>
--<br>
Brian West<br>
<a href="mailto:brian@freeswitch.org">brian@freeswitch.org</a><br>
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<div class="HOEnZb"><div class="h5"><br>
<br>
<br>
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On Feb 25, 2013, at 8:08 AM, Flavio Goncalves <<a href="mailto:flavio@sippulse.com">flavio@sippulse.com</a>> wrote:<br>
<br>
> Hi,<br>
><br>
> 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.<br>
><br>
> INVITE<br>
> a=rtpmap:18 G729/8000.<br>
> a=fmtp:18 annexb=yes.<br>
> a=rtpmap:0 PCMU/8000.<br>
> a=rtpmap:8 PCMA/8000.<br>
> a=rtpmap:101 telephone-event/8000.<br>
> a=fmtp:101 0-16.<br>
> a=rtpmap:19 CN/8000.<br>
><br>
> and FS is answering<br>
><br>
> 200OK<br>
> m=audio 31388 RTP/AVP 18 101 19.<br>
> a=rtpmap:18 G729/8000.<br>
> a=fmtp:18 annexb=yes.<br>
> a=rtpmap:101 telephone-event/8000.<br>
> a=fmtp:101 0-16.<br>
> a=rtpmap:19 CN/8000.<br>
> a=ptime:20.<br>
><br>
> If freeswitch supports only G729A, shouldn't FS refuse the call with 488 or suppress the line annexb=yes?<br>
><br>
> 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.<br>
><br>
> Flavio<br>
><br>
><br>
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