Oh, and it is allowed in the standard to override a static number with a rtpmap line for a dynamic codec if you run out of numbers in the dynamic range, but the unassigned numbers should be used before any of the assigned ones.<br>
<br>-Steve<br><br><br><br><div class="gmail_quote">On 16 January 2011 23:21, Steven Ayre <span dir="ltr"><<a href="mailto:steveayre@gmail.com">steveayre@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
It's part of the RTP/SDP standards. They're either static or dynamic numbers.<br><br>Dynamic ones (96-127) are named by name in a=rtpmap lines.<br><br>Static numbers are reserved numbers and the a=rtpmap is allowed but optional (although some devices incorrectly require it)<br>
The list of reserved static numbers is: <a href="http://www.iana.org/assignments/rtp-parameters" target="_blank">http://www.iana.org/assignments/rtp-parameters</a><br><br>-Steve<div><div></div><div class="h5"><br><br><br>
<div class="gmail_quote">On 16 January 2011 23:18, Diego Viola <span dir="ltr"><<a href="mailto:diego.viola@gmail.com" target="_blank">diego.viola@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">How do you understand these RTP/AVP numbers?<br>
<div><br>
m=audio 23344 RTP/AVP 3 18 98 99 9 0 8 101 13<br>
<br>
</div>What does the numbers mean? I don't see that on the wiki.<br>
<br>
Any help appreciated.<br>
<div><div></div><div><br>
On Sun, Jan 16, 2011 at 7:48 PM, David Ponzone <<a href="mailto:david.ponzone@ipeva.fr" target="_blank">david.ponzone@ipeva.fr</a>> wrote:<br>
> That sounds like a bad reading of the trace.<br>
> The important line is the 239th:<br>
> m=audio 23344 RTP/AVP 3 18 98 99 9 0 8 101 13<br>
> That means codec 3 is sent first (so GSM), then 18 (so G729).<br>
> I think you should read a little bit on early negotiation.<br>
> If leg A has GSM in 1st position, if your inbound codec list allows GSM, and<br>
> your outbound codec list is : G729, PCM, GSM, than your outbound INVITE to<br>
> leg B will have GSM in 1st position, because that was requested by leg A.<br>
> That's in the wiki, it's a part I rewrote myself some weeks ago, in the most<br>
> readable way I could.<br>
> David Ponzone Direction Technique<br>
> email: <a href="mailto:david.ponzone@ipeva.fr" target="_blank">david.ponzone@ipeva.fr</a><br>
> tel: 01 74 03 18 97<br>
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><br>
><br>
><br>
> Le 16/01/2011 à 21:59, Diego Viola a écrit :<br>
><br>
> Hello,<br>
><br>
> I'm experiencing some strange issue with codecs. I have the following<br>
> in my vars.xml file:<br>
><br>
> <X-PRE-PROCESS cmd="set"<br>
> data="global_codec_prefs=G729,G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/><br>
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,PCMA,GSM"/><br>
><br>
> "inbound-late-negotiation" and "disable-transcoding" are commented in<br>
> my internal SIP profile. So I guess I'm in Early Negotiation (default<br>
> behavior) mode.<br>
><br>
> However, when I send a call to my provider, and I look at the SIP<br>
> trace I see that FS is sending another codec, not G729 as I specified<br>
> in the global_codec_prefs / outbound_codec_prefs parameters.<br>
><br>
> I'm sending calls like this:<br>
><br>
> <action application="bridge" data="sofia/internal/$<a href="mailto:1@38.102.93.70" target="_blank">1@38.102.93.70</a>"/><br>
><br>
> Here is a SIP trace of a call:<br>
><br>
> <a href="http://pastebin.freeswitch.org/15042" target="_blank">http://pastebin.freeswitch.org/15042</a><br>
><br>
> I'm not understanding why FS is sending an INVITE with the G7221 codec<br>
> in line 240, if I'm telling it explicitly that I want G729 as the<br>
> priority when possible in the codec prefs options. But I see G729 in<br>
> the 200 OK in line 291.<br>
><br>
> I've been told to use absolute_codec_string=G729 in my dialplan or<br>
> enable late negotiation, but why if I'm already telling it to use G729<br>
> in the codec prefs?<br>
><br>
> my softphone IP: 190.23.80.10<br>
> provider IP: 38.102.93.70<br>
> FS IP: 77.92.65.126<br>
><br>
> calls flow like this:<br>
><br>
> softphone -> FS -> provider<br>
><br>
> Any help appreciated.<br>
><br>
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<br>
<br>
</div></div>--<br>
<div>Diego Viola<br>
Representative of Bridgecom LLC<br>
Phone: +595 971 320 520<br>
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<br>
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