On Tue, Aug 19, 2008 at 1:57 PM, Fabio Pietrosanti (naif) <lists at infosecurity.ch> wrote: > Hi Manoj, > > hooking directly inside pjmedia does not mean trowing away a lot of > pjsip code. > As far as i understood the biggest feature that will not be present > anymore will be the "conference calling" and some PCM level processing > (that i don't know exactly the requirement) > > Well basically almost *all* PJMEDIA features will be unavailable. The only things that are still usable are things related to transport, such as RTP/RTCP packetization, jitter buffer, and SRTP. > Maybe hooking inside pjmedia would also mean having the APS not only > provide the hardware-royalty-paid-for-free codecs, but that it would > also directly manage the Voice Activity Detection, that's a very > important feature in order to statistically reduce the bandwidth of 50% . > > Benny, Nanang am i right or not? > > Actually I'm not sure. We tested AMR with the built in SIP user agent in Nokia E90, and from Nokia there doesn't seem to be any VAD/DTX, while from pjmedia there is. With casual tests, the average bitrate from pjmedia is only about 2.6Kbps (since we use the 4.75Kbps mode), while from Nokia it's over 12Kbps. This doesn't say if VAD/DTX feature is not present, but at least it's not enabled by default, while in pjmedia it is. > I still did not analyzed in the details what would mean integrating APS > inside pjmedia even if it seems like the most "clean" design for Symbian > S60 integration, because directly leverage the platform capabilities. > > It shouldn't be too difficult to use device's codec with pjmedia, one just need to replace the stream implementation with a different one, leveraging the RTP/RTCP packetization and jitter buffer components that are already provided in pjmedia. Something like that. Cheers Benny -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20080819/fdae3b7d/attachment.html
