<div dir="ltr"><meta http-equiv="content-type" content="text/html; charset=utf-8"><span class="Apple-style-span" style="font-family: arial, sans-serif; font-size: 13px; border-collapse: collapse; "><p class="MsoNormal" style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">
<span class="Apple-style-span" style="font-size: small;">Anthony,</span></p><p class="MsoNormal" style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; "><span class="Apple-style-span" style="font-size: small;"><br>
</span></p><p class="MsoNormal" style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; "><span class="Apple-style-span" style="font-size: small;">This is what happened according to our log and wireshark </span><span class="Apple-style-span" style="font-size: small; ">analysis.</span></p>
<p class="MsoNormal" style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; "><span class="Apple-style-span" style="font-size: small; ">I will read through the RFC again see which side is taking it wrong (FS or PJSIP).</span></p>
<p class="MsoNormal" style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; "><span class="Apple-style-span" style="font-size: small; ">Either way, thank you and all others for time and energy you have spent answering this and other posts on the mailing list, let alone FreeSWITCH development and maintenance.</span></p>
<p class="MsoNormal" style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; "><span class="Apple-style-span" style="font-size: small; "><br></span></p><p class="MsoNormal" style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">
<span class="Apple-style-span" style="font-size: small; ">Tzury </span></p><p class="MsoNormal" style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; "><span class="Apple-style-span" style="font-size: small; "><br>
</span></p><p class="MsoNormal" style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; "><span class="Apple-style-span" style="font-size: small; "><br></span></p><p class="MsoNormal" style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">
<span class="Apple-style-span" style="font-size: small; ">Client caller say in the SDP speex uses PT=98.</span></p><p class="MsoNormal" style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; "><span class="Apple-style-span" style="font-size: small;">FS answer in SDP speex will use PT=98, and then the ring tone is played with PT=98 and the caller hears the ringback well.</span></p>
<p class="MsoNormal" style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; "><span class="Apple-style-span" style="font-size: small;"> </span></p><p class="MsoNormal" style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">
<span class="Apple-style-span" style="font-size: small;">FS invite destination with SDP speex PT=99 (value defined in the code).</span></p><p class="MsoNormal" style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">
<span class="Apple-style-span" style="font-size: small;">Client answers SDP speex uses PT=98.</span></p><p class="MsoNormal" style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; "><span class="Apple-style-span" style="font-size: small;"> </span></p>
<p class="MsoNormal" style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; "><span class="Apple-style-span" style="font-size: small;">Answer client sends speex with PT=99 (according the FS request) and caller send speex PT=98 (according to FS request).</span></p>
<p class="MsoNormal" style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; "><span class="Apple-style-span" style="font-size: small;">Answer waits to get speex with PT=98, and caller waits to get speex with PT=98 (the same it was with the ringtone).</span></p>
<p class="MsoNormal" style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; "><span class="Apple-style-span" style="font-size: small;">Answer side hears caller, but caller doesn't hear client.</span></p>
<p class="MsoNormal" style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; "><span class="Apple-style-span" style="font-size: small;"> </span></p><p class="MsoNormal" style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">
<span class="Apple-style-span" style="font-size: small;">If FS would have declare PT=(the value requested from the caller), the answer side would have send the PT equal to what was agreed with the FS first leg of the call.</span></p>
<div><span class="Apple-style-span" style="font-size: small;"><br></span></div></span><br><div class="gmail_quote">On Thu, Jul 8, 2010 at 7:54 PM, Anthony Minessale <span dir="ltr"><<a href="mailto:anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">The server is required to send the client on the payload it requested regardless of what the server chooses.<div><br>
</div>
<div><div>RFC 3264:</div><div> Once the answerer has sent the answer, it MUST be prepared to receive</div>
<div> media for any recvonly streams described by that answer. It MUST be</div><div> prepared to send and receive media for any sendrecv streams in the</div><div> answer, and it MAY send media immediately. The answerer MUST be</div>
<div> prepared to receive media for recvonly or sendrecv streams using any</div><div> media formats listed for those streams in the answer, and it MAY send</div><div> media immediately. When sending media, it SHOULD use a packetization</div>
<div> interval equal to the value of the ptime attribute in the offer, if</div><div> any was present. It SHOULD send media using a bandwidth no higher</div><div> than the value of the bandwidth attribute in the offer, if any was</div>
<div> present. The answerer MUST send using a media format in the offer</div><div> that is also listed in the answer, and SHOULD send using the most</div><div> preferred media format in the offer that is also listed in the answer.</div>
<div> <span style="background-color:rgb(255, 255, 51)">In the case of RTP, it MUST use the payload type numbers</span></div><div><span style="background-color:rgb(255, 255, 51)"> from the offer, even if they differ from those in the answer.</span></div>
</div><div><br></div><div><br></div><div>What is pretty amusing though is from the same RFC:</div><div><br></div><div><br></div><div><div> In the case of RTP, if a particular codec was referenced with a</div><div> specific payload type number in the offer, that same payload type</div>
<div> number SHOULD be used for that codec in the answer. Even if the same</div><div> payload type number is used, the answer MUST contain rtpmap</div><div> attributes to define the payload type mappings for dynamic payload</div>
<div> types, and SHOULD contain mappings for static payload types. The</div><div> media formats in the "m=" line MUST be listed in order of preference,</div><div> with the first format listed being preferred. In this case,</div>
<div> preferred means that the offerer SHOULD use the format with the</div><div> highest preference from the answer.</div></div><div><br></div><div><br></div><div>So SHOULD means they have no nerve to enforce it because it's ambiguous and we can choose not to do it.</div>
<div>The MUST in the first example is what we follow.</div><div><br></div><div>MAYBE its possible to find a way to prefer the callers pt like we SHOULD, we'll have to think about it.</div><div><br></div><div><br></div>
<div><br></div><div><br></div><div><div><div></div><div class="h5"><br><br><div class="gmail_quote">On Thu, Jul 8, 2010 at 9:28 AM, Tzury Bar Yochay <span dir="ltr"><<a href="http://tzury.by" target="_blank"><font class="Apple-style-span" color="#000000">tzury.by</font></a>@<a href="http://reguluslabs.com" target="_blank"><font class="Apple-style-span" color="#000000">reguluslabs.com</font></a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div>Brian and Anthony,</div><div>Dears,</div><div><br></div><div>I am afraid it is a bug in teh FS</div>
<div>See, if during SDP the client declare on payload 98 and fs declare on 99.</div>
<div>Then client expect receiving 99 while sending 98. That is fine.</div><div>However, when both clients are the same (as in our case) since FS is not trans-coding</div><div>clients receives 98 in</div><div>given that same sip client cannot work with previous versions of FS</div>
<div>where there Speex was 98 (and 97 in older) </div><div><div></div><div><div><br></div><br><div class="gmail_quote">On Thu, Jul 8, 2010 at 5:08 PM, Anthony Minessale <span dir="ltr"><<a href="mailto:anthony.minessale@gmail.com" target="_blank"><font class="Apple-style-span" color="#000000">anthony.minessale@gmail.com</font></a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><p>No, you can't force codecs in the dynamic range and if properly implemented, it won't matter what number either side chooses.</p>
<p></p><blockquote type="cite"><div><div></div><div>On Jul 8, 2010 8:34 AM, "Tzury Bar Yochay" <<a href="http://tzury.by" target="_blank"><font class="Apple-style-span" color="#000000">tzury.by</font></a>@<a href="http://reguluslabs.com" target="_blank"><font class="Apple-style-span" color="#000000">reguluslabs.com</font></a>> wrote:<br>
<br>Hi,<br>
<br>
I just download the latest, build and brought up all is fine.<br>
However, our clients are connected via IP over GPRS(UMTS) and thus we<br>
are using Speex codec.<br>
<br>
Is there a way I can enforce speex to have 97 or 98 instead of 99?<br>
Currently as it looks in the vars.xml speex is 99.<br>
see dump (taken from default vars.xml)<br>
<br>
thanks<br>
Tzury<br>
RTP Dynamic Payload Numbers currently used in FreeSWITCH and what for.<br>
<br>
96 - AMR<br>
97 - iLBC (30)<br>
98 - iLBC (20)<br>
99 - Speex 8kHz, 16kHz, 32kHz<br>
100 -<br>
<br></div></div><div>
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