symbian_ua_gui with TLS, VAS and G729 protocol via SRTP proxy

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Dmitry-

> Jeff, do you mean that operator may impair packets on edge and leave
> intact switching to umts?

Yes if the data packets are RTP packets (and possibly SIP, I've heard from some customers lately that a few carriers
are using more sophisticated blocking algorithms).

But again, only some carriers might do it.  You could be having some other problem.  This is why I suggest
Wiresharking so you can post debug samples of what you're seeing -- maybe some packets out of sequence, maybe some
missing, etc.  It's hard to guess on something like this.

-Jeff

> On 5/10/10, Jeff Brower <jbrower at signalogic.com> wrote:
>> Dmitry-
>>
>>> 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.
>>
>> And the other main problem is that many carriers block (or impair) voice
>> transport over non-voice channels.  In that
>> case you have to encrypt or otherwise "hide" RTP (and sometimes SIP) packets
>> on data channels.  Also when you say you
>> can "hear a packet" that's impossible because you're talking about 10 or 20
>> msec length of time -- less than 1/50th of
>> a second.
>>
>> My suggestion would be to put Wireshark on either side of Kamailio and find
>> out what is actually happening to your
>> packets.  If it's a deliberate impairment you should be able to tell.
>>
>> -Jeff
>>
>>> 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.
>>
>>
>
> --
> Sent from my mobile device
>




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