[Freeswitch-users] Change ptime value

[David Witham]

If you're already passing G729 through, including G729 at 40i​ in the list should override the ptime. Check the SDP on the outbound leg of the call to confirm.


________________________________
From: freeswitch-users-bounces at lists.freeswitch.org <freeswitch-users-bounces at lists.freeswitch.org> on behalf of bhavik patel <bhavikpatel14388 at gmail.com>
Sent: Monday, 24 August 2015 17:29
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Change ptime value

Is it possible i can change it for G729 codec ?
Like G729 at 40i ?

On my Server G729 as Pass-through mode.


On Mon, Aug 24, 2015 at 6:40 AM, David Witham <david.witham at netsip.com.au<mailto:david.witham at netsip.com.au>> wrote:

Hi Bhahvik,


We have instances were ptime:30 is signalled but the stream is actually ptime:20 so we override the signalling with this:


absolute_codec_string=PCMA at 20i,PCMU at 20i​


It is slightly different to your examples below but 3 and 4 are close - I've used @ instead of "at".


Hope this helps,

David

________________________________
From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> <freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>> on behalf of Kamil Nigmatullin <kamil.nigmatullin at gmail.com<mailto:kamil.nigmatullin at gmail.com>>
Sent: Saturday, 22 August 2015 22:57
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Change ptime value


I know you can do this easyly with opensips

22 авг. 2015 г. 16:10 пользователь "bhavik patel" <bhavikpatel14388 at gmail.com<mailto:bhavikpatel14388 at gmail.com>> написал:
Hello Everyone,

I want to change ptime configuration. Will anyone  assist me to change ptime value 20 to 40 for codec PCMU,PCMA ?

FLow : SIP-PHONE---Ptime(20)-->FS server (Need to change Ptime to 40) ---------->To Provider

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.

Currently I can see below log in freeswitch logs :

m=audio 8000 RTP/AVP 98 97 8 0 3 101
a=rtpmap:98 speex/16000
a=rtpmap:97 speex/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
Dialplan: sofia/default/4602534812 at 192.168.1.26<mailto:4602534812 at 192.168.1.26> Action bridge(sofia/gateway/localgateway/001***)

m=audio 24716 RTP/AVP 8 101 13
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

I tried different solutions for this issue but no luck.

1. Add below line before Bridge Channel
  set(absolute_codec_string=PCMU at 8000@40i,PCMA at 8000@40i)

2. Add below line before Bridge Channel
 set(absolute_codec_string=PCMU at 8000 at 40i,PCMA at 8000 at 40i)

3.Change below parameter in SIP profile
   1.outbound-codec-prefs=PCMU,PCMA to outbound-codec-prefs=PCMU at 8000@@40i,PCMU at 16000@@40i,PCMU at 32000@@40i,PCMA at 8000@40i,PCMA at 16000@40i,PCMA at 32000@40i

   2.outbound-codec-prefs=PCMU,PCMA,rtp-autofix-timing=false

   3.outbound-codec-prefs=PCMU,PCMA,absolute_codec_string=PCMU at 40i,PCMA at 40i,rtp-autofix-timing=false

   4.outbound-codec-prefs=PCMU at40i,PCMA at 40i

   5.outbound-codec-prefs=PCMU at 8000h at 40i,PCMA at 8000h at 40i,inbound-late-negotiation=false

   6.outbound-codec-prefs=G729,PCMU,PCMA,inbound-late-negotiation=false


Please point me where i am doing wrong.


--
Thanks,
Bhavik Patel


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--
Thanks,
Bhavik Patel

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