21 Aug
2005
21 Aug
'05
7:03 a.m.
That does not help you - inserting another hop can not compensate the
bottleneck between the UAs and the Internet (the ADSL connection).
BTW: 600km is not far away. Which Codec you you use? Try low-babdwidth
codecs like G729, ilbc, GSM and G723
klaus
Ozan Blotter wrote:
Dear Iqbal and Bogdan,
That is exactly what i need to improve QoS, because the UAs ADSL
connection is not fast enough, 512/128k in general, OpenSER and UA's are
located in same city but PSTN GW is nearly 600 km (350 miles nearly) away
from this point. I thought since OpenSER is on a good connection (1Mbit
Fiber) and can relay to PSTN faster then UA's themselves.
Thanks,
Ozan
Also in terms of ping times how far is the PSTN gateway, as Bogdan said,
media proxy will add even more delay, as for sending pstn via
mediaproxy, you could do all yourprocessing, and once ser detects that
call must goto PSTN, then do all the NAT checking ...this is a
oversimplified view...
Iqbal
Bogdan-Andrei Iancu wrote:
Hi Ozan,
so, if I get it right, you have a PSTN gw locate far away from the
UAs...and instead of letting the media traffic to do directly between
GW and UAs, you want to insert in the middle a RTP relay (mediaproxy)
in order to improve the QoS, right?
If so, I would say this will not help you - having a media relay on
your media path leads to QoS degradation due additional delays....
If you still want to go for it....I will take a look on tour script.
regards,
bogdan
Ozan Blotter wrote:
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Dear List,
At below config. i setted up OpenSER on Public IP and PSTN Gateway is
again on another Public IP. I use Zyxel Prestige Series 660-HW ADSL
modems on both client side and note that they have SIP ALG, also i've
completed port forwarding in each modem, so SIP2SIP call happens
(voice traffic goes from UA to UA) without need of anything extra as
STUN or RTP RELAY. But since those are very away from each other if a
UA needs to call PSTN the voice traffic flows from UA to PSTN and
that lowers QoS :(
I would like to activate MediaProxy again but just for one case which
is PSTN, so whenever a UA needs to call PSTN the voice traffic flows
like :
UA (ATA) >>> OpenSER + MediaProxy >>> PSTN
How can i setup this like above without touching SIP2SIP calls
between UA's...
or adding the lines beginning with # these to my existing config is
enough ?
Thanks,
Ozan Blotter
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