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<p>Hi Bhahvik,<br>
</p>
<p><br>
</p>
<p>We have instances were ptime:30 is signalled but the stream is actually ptime:20 so we override the signalling with this:<br>
</p>
<p><br>
</p>
<p>absolute_codec_string=PCMA@20i,PCMU@20i​<br>
</p>
<p><br>
</p>
<p>It is slightly different to your examples below but 3 and 4 are close - I've used @ instead of "at".<br>
</p>
<p><br>
</p>
<p>Hope this helps,<br>
</p>
<p>David<br>
</p>
<div style="color: rgb(33, 33, 33);">
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<div id="divRplyFwdMsg" dir="ltr"><font face="Calibri, sans-serif" color="#000000" style="font-size:11pt"><b>From:</b> freeswitch-users-bounces@lists.freeswitch.org <freeswitch-users-bounces@lists.freeswitch.org> on behalf of Kamil Nigmatullin <kamil.nigmatullin@gmail.com><br>
<b>Sent:</b> Saturday, 22 August 2015 22:57<br>
<b>To:</b> FreeSWITCH Users Help<br>
<b>Subject:</b> Re: [Freeswitch-users] Change ptime value</font>
<div> </div>
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<div>
<p dir="ltr">I know you can do this easyly with opensips </p>
<div class="gmail_quote">22 авг. 2015 г. 16:10 пользователь "bhavik patel" <<a href="mailto:bhavikpatel14388@gmail.com">bhavikpatel14388@gmail.com</a>> написал:<br type="attribution">
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex; border-left:1px #ccc solid; padding-left:1ex">
<div dir="ltr">
<div>Hello Everyone,<br>
<br>
I want to change ptime configuration. Will anyone assist me to change ptime value 20 to 40 for codec PCMU,PCMA ?<br>
<br>
</div>
FLow : <b>SIP-PHONE---Ptime(20)-->FS server (Need to change Ptime to 40) ---------->To Provider</b><br>
<div><br>
</div>
<div>NOTE : If I send 40 Ptime From SIP-PHONE then Call quality is Good but for this i need to change all phones' ptime and I don't want to change it manually.So I need to change this From FS Side.<br>
</div>
<div><br>
Currently I can see below log in freeswitch logs :<br>
<br>
m=audio 8000 RTP/AVP 98 97 8 0 3 101<br>
a=rtpmap:98 speex/16000<br>
a=rtpmap:97 speex/8000<br>
a=rtpmap:8 PCMA/8000<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:3 GSM/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-15<br>
a=ptime:20<br>
Dialplan: sofia/default/<a href="mailto:4602534812@192.168.1.26" target="_blank">4602534812@192.168.1.26</a> Action bridge(sofia/gateway/localgateway/001***)<br>
<br>
m=audio 24716 RTP/AVP 8 101 13<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=ptime:20<br>
a=sendrecv<br>
<br>
I tried different solutions for this issue but no luck.<br>
<br>
1. Add below line before Bridge Channel<br>
set(absolute_codec_string=PCMU@8000@40i,PCMA@8000@40i)<br>
<br>
2. Add below line before Bridge Channel<br>
set(absolute_codec_string=PCMU at 8000@40i,PCMA at 8000@40i)<br>
<br>
3.Change below parameter in SIP profile<br>
1.outbound-codec-prefs=PCMU,PCMA to outbound-codec-prefs=PCMU@8000@@40i,PCMU@16000@@40i,PCMU@32000@@40i,PCMA@8000@40i,PCMA@16000@40i,PCMA@32000@40i<br>
<br>
2.outbound-codec-prefs=PCMU,PCMA,rtp-autofix-timing=false<br>
<br>
3.outbound-codec-prefs=PCMU,PCMA,absolute_codec_string=PCMU at 40i,PCMA at 40i,rtp-autofix-timing=false<br>
<br>
4.outbound-codec-prefs=PCMU at40i,PCMA at 40i<br>
<br>
5.outbound-codec-prefs=PCMU at 8000h@40i,PCMA at 8000h@40i,inbound-late-negotiation=false<br>
<br>
6.outbound-codec-prefs=G729,PCMU,PCMA,inbound-late-negotiation=false<br>
<br>
<br>
Please point me where i am doing wrong.<br clear="all">
<br>
<br>
-- <br>
<div>Thanks,<br>
Bhavik Patel<br>
<br>
</div>
</div>
</div>
<br>
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