After some evaluation of bandwidth requirements for Mobile VoIP it appeared very clear to me that the standard RTP cause really, a lot of overhead, in the ip transport. In order to reduce the bandwidth requirement of an RTP stream several proposal appeared like CRTP (http://www.faqs.org/rfcs/rfc3545.html) and ROHC (http://en.wikipedia.org/wiki/ROHC) . ROHC is the winner of the standard for RTP header compression and will even be used in 4G telephony. BUT those standards are only designed for point-to-point connections. This means that they can do compression over a point-to-point channel like the one between the mobile-phone and the BTS, or between two endpoints connected with a dedicated E1/T1 line. However those standards are not good for generic usage over today, commonly available, ip networks. Fring, talkonaut and a lot of other mobile voip software implement a cutted-down version of RTP in order to stay even within the GPRS bandwidth requirements. It's simply a matter of reducing the 12byte RTP header to something more *pragmatic* for mobile use. My questions is: A standard for such kind of optimization exists? To me appear not to exists. In XMPP world a standard very efficient appeared: NO HEADER, raw udp transport: http://www.xmpp.org/extensions/xep-0177.html If it does not exists, does it could be possible to work on something like a reference implementation? Are there other companies interested in having a mobile voip client with very narrowband requirements that can work even on GPRS like Fring and Talkonaut by creating a stripped down version of RTP? From an implementation point of view, unfortunately, it seems to me that this kind of protocol (stripped down minimalistic RTP) must be implemented not only on the client side (pjsip) but also on a server side with some kind of proxying technology: VoIP client with stripped RTP <----> RTP (stripped) to RTP (non-stripped) proxy <---> RTP gateway (Asterisk, Yate or whatever) What the community think about it? Fabio
