[Freeswitch-users] Change ptime value

[bhavik patel]

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>
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 <
> freeswitch-users-bounces at lists.freeswitch.org> on behalf of Kamil
> Nigmatullin <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> 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 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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>
> _________________________________________________________________________
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> http://www.freeswitchsolutions.com
>
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>
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-- 
Thanks,
Bhavik Patel
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