Hello,
As far as I could understand, I was able to configure and
compile symbian_ua_gui with TLS, VAS and G729 protocol to work with local
Kamailio based server, all communications go through media-proxy, SRTP (on
the same local server).
In 3G mobile networks, everything works just fine and quality is superb.
Now what I see when connecting two entities (Nokia E51) WHEN CONNECTED
THROUGH EDGE (the place where both entities were located has *no* 3G access
and this was simulated by switching the phones) there is a very bad quality
of audio.
Either the packets are too big for such a slow connection (you can mostly
here about a half of each packet then overlapped by the next one) or my
configuration lacks some other vital parameters. It is not like the behavior
we hear in GSM connection ("robotizing", "bubbling") - just packets are
overlapped.
In general, the main problem with mobile networking is that you have a
rather wide Inbound channel and very narrow Outbound channel. Thus, if at
least one entity has wide narrowband (3G, WiFi connected), the other side
have clean incoming audio. The tests of the mobile network are the
following:
1st entity: 0,22Mbps incoming, 0,07Mbps outgoing.
2nd entity: 0,15Mbps incoming, 0,02Mbps outgoing.
According to at least this table (
http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a0080094ae2.shtml)
the link has enough bandwidth to serve one connection. Even 9Kbps each
direction should be enough.
The questions are:
1. Whether anyone tested symbian_ua_gui or symbian_ua in EDGE phone mode,
not 3G or WiFi?
2. What could I check in configuration of pjmedia, pjsip in general and
symbian_ua/_gui in particular to make it work on EDGE?
Thanks in advance,
Dmitry.
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