Is it possible i can change it for G729 codec ? Like G729@40i ? On my Server G729 as Pass-through mode. On Mon, Aug 24, 2015 at 6:40 AM, David Witham <david.witham@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@20i,PCMU@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@lists.freeswitch.org < freeswitch-users-bounces@lists.freeswitch.org> on behalf of Kamil Nigmatullin <kamil.nigmatullin@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@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@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@8000@40i,PCMA@8000@40i)
2. Add below line before Bridge Channel set(absolute_codec_string=PCMU at 8000@40i,PCMA at 8000@40i)
3.Change below parameter in SIP profile 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
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@40i,PCMA at 8000h@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