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? >> -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20100201/7f60282f/attachment.html>
