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.
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Thanks,
Bhavik Patel