Hi, Well, as remote specifies mode-set param, the local encoder must obey it, rfc4867 sec 8.1: "... If mode-set is specified, it MUST be abided, ...." Regarding jitter buffer, I think we'll always need it for 'normalizing' jitter and clock drift, so instead of omitting it, perhaps it's better to improve it and share the good result to the community :) BR, nanang 2010/2/1 cq_wei <cq_wei at 126.com>: > naning, > It does not sound change anything. The 200 OK message have SDP setting > modeset=1 which means 5150bps is always selected, > so the fix for ticket #1028 does not change anything. For ticket #969, Would > passthrough codec using jitter buffer of pjmedia? I suppose we should > minimize jitter buffer for passthrough codec and let the rtp packet > passthrough to VAS directly! > > I have merge both fix but the voice quality still have gap with Nokia's own > implementation. > > Anyway thanks for help. I will dive deeper into code to trace the problem. > > > >> >Message: 1 >Date: Fri, 29 Jan 2010 19:45:53 +0700 >> > >From: Nanang Izzuddin <nanang@xxxxxxxxx> >> > >To: pjsip list <pjsip at lists.pjsip.org> >> > >Subject: Re: VAS/AMR codec fine tuning problem for GPRS/EDGE > >> > network >Message-ID: > >> > <f8a01ced1001290445w129d3374x4e3ad1225dbd3f6d at mail.gmail.com> >> > >Content-Type: text/plain; charset=GB2312 > >Hi, > >> > >Please update your source from SVN trunk, in case you haven't. There >> > >was a bug in VAS wrapper regarding bitrate setting, so AMR always >> > >worked on 4750bps bitrate regardless the setting. This has been fixed >> > >in changeset 3078 (of ticket #1028). Also, there has been a work on >> > >jitter buffer for low bandwidth network, please see ticket #969. > >> > >We'll be glad to hear any feedbacks. > >BR, >nanang > > >> > >2010/1/29 cq_wei <cq_wei at 126.com>: >> Hi all, >> >> > >> Anybody know if there is more params to adjust? I am building an user agent >> > >> on symbian platform using VAS/AMR codec, it works but the voice quality is >> > >> poor. In order to fine tuning I have change the following things >> >> > >> 1) Default codec , I have changed the definition for AMR's codec_desc >> > >> in passthrough codec.c. the default mode and frame per packet have been >> > >> adjusted (I just change them same as nokia's own PoC client). >> >> > >> 2) Media_cfg. Using default configuration. default param is most >> > >> suitable for VAS/AMR passthrough codec, right? >> >> > >> Still the quality is uncomparable with Nokia's own client (NOKIA PTT >> > >> client). The delay is longer (Nokia's is 1 second around while mine is 2 >> > >> seconds around) and jitter is obvious. The network is GRPS/EDGE. >> >> > >> Is there any more params I can adjust to improve voice quality? >> > > ________________________________ > ??????????????????? > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > >
