I use dialup connection with my IPPhones which has a buildin modem without any issue on both codec G723 and G729. I sometime get better quality on the some phone if I use dialup then when I was braodband. this happens when I call most aftrican country where the termination is not good. This is because when I use broadband the media has to through some kinds of NAT transval while the dialup normally gets public IP so no NAT transval and the media gets the shorts path.
thanks
Mohamed
Jorge Crichigno <jcrichigno(a)conexion.com.py> wrote:
I think the data rates with overhead are as follow:
----------------------------------------------------------------
iLBC - 30 ms (a packet each 30 ms)
RTP payload: 50 bytes --> Rate: 13.3 Kbps
RTP: 62 bytes
UDP: 70 bytes
IP: 90 bytes --> total rate: 24 Kbps
-------------------------------------------------------------------
iLBC - 20 ms (a packet each 20 ms)
RTP payload: 38 bytes --> Rate: 15.2 Kbps
RTP: 50 bytes
UDP: 58 bytes
IP: 78 bytes --> total rate: 31.2 Kbps
-------------------------------------------------------------------
G.729 - 2 voice frame per packet
RTP payload: 20 bytes --> Rate: 8 Kbps
RTP: 32 bytes
UDP: 40 bytes
IP: 60 bytes --> Total rate: 24 Kbps
-------------------------------------------------------------------
G.729 - 4 voice frame per packet
RTP payload: 40 bytes --> Rate: 8 Kbps
RTP: 52 bytes
UDP: 60 bytes
IP: 80 bytes --> Total rate: 16 Kbps
-------------------------------------------------------------------
G.711 - 20 ms (a packet each 20 ms)
RTP payload: 160 bytes --> Rate: 64 Kbps
RTP: 172 bytes
UDP: 180 bytes
IP: 200 bytes --> Total rate: 80 Kbps
El vie, 27-05-2005 a las 10:36, Iqbal escribió:
> the 5.3, 6.3K are really theoretical, i dont think they include IP
> overheads, I used media_sessions.phtml, and looked at the actual calls
> per codec, and I dont think u can really get a good call without 50-70K,
> also most bandwidth providers (at least here in the UK) are
> asymmetrical, so even on a 128K, u could have problems.
>
> Having said that I have done a nice call on xlite using ilbc on dial up.
>
> Iqbal
>
> Kofi Obiri-Yeboah wrote:
>
> >Just to add a few more details, Greger is right to point out the quality
> >inferiority of G.723 compared to those of G711 and G729. In fact, in most VOIP
> >deployments, in order to quarantee interoperability, a minimum bandwidth of
> >128K is specified. However to reach the wider "lower bandwidth areas" most
> >service providers are opting for G.723 which uses either 5.3 or 6.3K. At this
> >low bandwidth transmission needs, one could literally reach "dial up modem"
> >equipped areas. in fact most VOIP phone hardware and software are begining to
> >specify G.723 as their default codec. Note that until the direct media
> >connection phase of a VOIP vall setup, wide bandwidth is not required. Also
> >note that analogue phones have a maximim bandwidth need of 3K, hence even the
> >low quality of G.723/5.3K, compared to the average analogue phone call, is
> >superior
> >
> >
>
> _______________________________________________
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> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
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Hi All,
I really need your help.
I have setup the ser-0.8.14,
uname �a is:
Linux lnx01 2.6.9-1.667smp #1 SMP Tue Nov 14:59:52 EST 2004 i686 i386 GNU/Linux
I have also setup Radiusclient-0.4.8 and Freeradius-1.0.2.
I've followed all the steps in this tutorial
http://www.iptel.org/ser/doc/ser_radius/ser_radius.html
The installing don�t have any error.
I tested on LAN with 1 server 192.168.10.1 and 2 workstations 192.168.10.2 and 192.168.10.3
However, I have received error when start ser.
This is my error:
0(7389) find_export: found <db_update> in module mysql [/usr/local/lib/ser/modules/mysql.so]
0(7389) ERROR: acc: can't get code for the Failed attribute value
0(7389) init_mod(): Error while initializing module acc
ERROR: error while initializing modules
0(7389) DEBUG: tm_shutdown : start
0(7389) DEBUG: tm_shutdown : empting DELETE list
I should be most grateful to receive your help.
Best regards,
Tran Tien Duc
ser.cfg
debug=8
fork=no
log_stderror=yes
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
#port=5060
#children=4
fifo="/tmp/ser_fifo"
# ------------------ module loading ----------------------------------
# Uncomment this if you want to use SQL database
loadmodule "/usr/local/lib/ser/modules/mysql.so"
loadmodule "/usr/local/lib/ser/modules/sl.so"
loadmodule "/usr/local/lib/ser/modules/tm.so"
loadmodule "/usr/local/lib/ser/modules/rr.so"
loadmodule "/usr/local/lib/ser/modules/maxfwd.so"
loadmodule "/usr/local/lib/ser/modules/usrloc.so"
loadmodule "/usr/local/lib/ser/modules/registrar.so"
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
loadmodule "/usr/local/lib/ser/modules/auth.so"
loadmodule "/usr/local/lib/ser/modules/acc.so"
loadmodule "/usr/local/lib/ser/modules/auth_db.so"
loadmodule "/usr/local/lib/ser/modules/auth_radius.so"
loadmodule "/usr/local/lib/ser/modules/uri_radius.so"
loadmodule "/usr/local/lib/ser/modules/group_radius.so"
# ----------------- setting module-specific parameters ---------------
# -- usrloc params --
#modparam("usrloc", "db_mode", 0)
# Uncomment this if you want to use SQL database
# for persistent storage and comment the previous line
modparam("usrloc", "db_mode", 2)
modparam("usrloc","db_url","mysql://ser:heslo@localhost/ser")
# -- auth params --
# Uncomment if you are using auth module
#
modparam("auth_db", "calculate_ha1", yes)
#
# If you set "calculate_ha1" parameter to yes (which true in this config),
# uncomment also the following parameter)
#
modparam("auth_db", "password_column", "password")
# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)
modparam("acc","db_url","mysql://ser:heslo@localhost/ser")
modparam("acc","db_flag",1)
modparam("acc","log_level",1)
modparam("acc","log_flag",1)
modparam("acc","radius_config","/usr/local/etc/radiusclient/radiusclient.conf")
modparam("acc","radius_flag",1)
modparam("acc","radius_missed_flag",2)
# ------------------------- request routing logic -------------------
# main routing logic
route{
# initial sanity checks -- messages with
# max_forwards==0, or excessively long requests
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
break;
};
if ( msg:len > max_len ) {
sl_send_reply("513", "Message too big");
break;
};
# we record-route all messages -- to make sure that
# subsequent messages will go through our proxy; that's
# particularly good if upstream and downstream entities
# use different transport protocol
record_route();
# loose-route processing
if (loose_route()) {
t_relay();
break;
};
setflag(1);
if (method=="INVITE")
record_route();
# forward to current uri now; use stateful forwarding; that
# works reliably even if we forward from TCP to UDP
if (!t_relay()) {
sl_reply_error();
break;
};
}
---------------------------------
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Hi,
I have an Asterisk box up and running behind a NAT
(Local IP Address 192.168.1.250). I would like to have
another Linux box with SER as a proxy/registrar server
on the same NAT (Local IP Address 192.168.1.150) to
have a remote phone on ANOTHER NAT to register to the
Asterisk server.
My Questions are:
SER Server:
- Which ser.cfg shall I use, the default cfg file or
any other cfg on the examples folder?
- Do I have to have a Domain?
- Do I have to Port Forwarding 5060 to the SER box?
Asterisk Box:
- What are the changes required?
Client (Phone):
- Do I have to use Outbound Proxy and point it to the
WAN IP Address of the NATs Router?
- Any other special configurations required?
Any help is highly appreciated.
Regards.
__________________________________________________
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thank you very much
but it didn't work too
it's crazy ;)
if someone else already used what i'm looking fo (serctl fifo t_uac_dlg ...)
thanks
M.
>From: "Samuel Osorio Calvo" <samuel.osorio(a)nl.thalesgroup.com>
>To: <ladia6(a)centrum.cz>, <mikep3000(a)hotmail.com>, <serusers(a)lists.iptel.org>,
><greger(a)teigre.com>
>Subject: Re: [serusers] fifo
>Date: Thu, 26 May 2005 10:49:17 +0200
>
>Hi!
>
>I have never used the command line to use the t_uac_dlg fifo command, but
>just a script similar to the example present in the attached file FIFO.txt
>(it is from an older version of SER but the FIFO interface has not
>changed).
>I also attach another document containing the explanation found in the
>source code about the FIFO interface.
>
>I hope with these two documents you will be able to send SIP requests with
>t_uac_dlg
>
>Samuel.
>
>
>Unclassified.
> >>> "michael p" <mikep3000(a)hotmail.com> 05/25/05 04:32PM >>>
>i'm sorry but it didn't help me
>
>i think it's different with serctl but thanks
>
>i don't understand anybody never used this command line in serctl???
>
>thanks
>
>
>
>
> >From: "Greger V. Teigre" <greger(a)teigre.com>
> >To: "michael p" <mikep3000(a)hotmail.com>, <ladia6(a)centrum.cz>,
> ><serusers(a)lists.iptel.org>
> >Subject: Re: [serusers] fifo
> >Date: Wed, 25 May 2005 16:26:44 +0200
> >
> >Maybe this can help you in the right direction:
> >http://www.onsip.org/modules/xoopsfaq/index.php?cat_id=3#q2
> >g-)
> >
> >michael p wrote:
> >>only in the third link ther is my question but there is no answer
> >>Does anyone know what correct format of a packet
> >>should be pushed into fifo buffer? For example,
> >>
> >>serctl fifo t_uac_dlg BYE sip:1111 at xxx.xxx.xxx
> >>'sip:from:1111 at xxx.xxx.xxx' 'sip:to:2222 at xxx.xxx.xxx'
> >>'callid:xxxxxxxxxxxx' 'Cseq:xxxxxx' . .
> >>The above command I've tried,but got errors.
> >>
> >>
> >>and i tried the format into the serctl too
> >>
> >>
> >>thanks for replying
> >>
> >>
> >>
> >>>From: Ladislav Andel <ladia6(a)centrum.cz>
> >>>Reply-To: Ladislav Andel <ladia6(a)centrum.cz>
> >>>To: "michael p" <mikep3000(a)hotmail.com>
> >>>Subject: Re: [serusers] fifo
> >>>Date: Wed, 25 May 2005 15:39:07 +0200
> >>>
> >>>I'm sorry, I think I've seen it in the list, but can't find it..
> >>>
> >>>try the links below
> >>>
> >>>http://lists.iptel.org/pipermail/serusers/2004-August/010850.html
> >>>http://lists.iptel.org/pipermail/serusers/2004-August/010825.html
> >>>http://lists.iptel.org/pipermail/serusers/2005-April/018897.html
> >>>
> >>>mp> thanks but it didn't help me cause this topic not found in
> >>>google in this
> >>>mp> name maybe another one
> >>>
> >>>mp> so if you know which line i have to write it will be more simple
> >>>
> >>>mp> thanks
> >>>
> >>>
> >>>
> >>>>>From: Ladislav Andel <ladia6(a)centrum.cz>
> >>>>>Reply-To: Ladislav Andel <ladia6(a)centrum.cz>
> >>>>>To: "michael p" <mikep3000(a)hotmail.com>
> >>>>>CC: serusers(a)lists.iptel.org
> >>>>>Subject: Re: [serusers] fifo
> >>>>>Date: Wed, 25 May 2005 14:55:45 +0200
> >>>>>
> >>>>>This topic was in the list recently. Try to google it out.
> >>>>>Lada
> >>>>>
> >>>>>mp> i already saw the different options but i want to know how to
> >>>>>use them to
> >>>>>mp> send a message like INVITE or OPTIONS i don't know how i have
> >>>>>to write the
> >>>>>mp> commande
> >>>>>
> >>>>>mp> thanks
> >>>>>
> >>>>>>>From: Llanos Serna García-Conde <llanosserna(a)hotmail.com>
> >>>>>>>To: mikep3000(a)hotmail.com
> >>>>>>>CC: serusers(a)lists.iptel.org
> >>>>>>>Subject: RE: [serusers] fifo
> >>>>>>>Date: Wed, 25 May 2005 14:20:27 +0200
> >>>>>>>
> >>>>>
> >>>>>
> >>>
> >>>
> >>>
> >>
> >>
> >>_______________________________________________
> >>Serusers mailing list
> >>serusers(a)lists.iptel.org
> >>http://lists.iptel.org/mailman/listinfo/serusers
> >
>
>
>_______________________________________________
>Serusers mailing list
>serusers(a)lists.iptel.org
>http://lists.iptel.org/mailman/listinfo/serusers
>
><< FIFO.txt >>
><< FIFO_source.txt >>
Hi,
When i use "serctl moni", it shows,
Transaction Statistics
Current: 2 (47 waiting) Total: 19299 (12 local)
I understand that ser keeps the sessions for 5 seconds before
destroying them. But I saw the number of waiting increasing during the
last few days.
Do it mean that there is a memory leak or something?
Thanks,
Richard
Hello List:
between G723.1 and G729, what codec do you prefer?
I am just trying to understand pros and cons of both codec.
Thanks,
MOhammad
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Just curious to know,
Does this codec support depends on the actual hardware or the firmware?
MOhammad
Original Message:
-----------------
From: Juha Heinanen jh(a)tutpro.com
Date: Fri, 27 May 2005 17:48:36 +0300
To: info(a)beeplove.com, jcrichigno(a)conexion.com.py, serusers(a)iptel.org
Subject: Re: [Serusers] G723.1 vs G729
info(a)beeplove.com writes:
> Do you guys know about any hardphone that support Speex and iLBC codec?
> Most of the hardphone in market support G723, G711 and G729.
grandstream and snom, for example, have iLBC support. i have been told
that iLBC is also on sipura roadmap, but who knows what will happen to
that after the cisco deal. i have also heard rumors that iLBC support
is coming to cisco ios, mainly because iLBC was chosen for cable tv
terminals.
-- juha
--------------------------------------------------------------------
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http://mail2web.com/ .
I tried G723 on hardphone on dialup over mediaproxy.
On Dialup UA end up/down stream was 24k/19k
It was nice call but sometimes both party of both end noticed voice loss.
Thanks,
MOhammad
Original Message:
-----------------
From: Iqbal iqbal(a)gigo.co.uk
Date: Fri, 27 May 2005 15:36:07 +0100
To: kofi(a)radiocomplex.com, serusers(a)iptel.org
Subject: Re: [Serusers] G723.1 vs G729
the 5.3, 6.3K are really theoretical, i dont think they include IP
overheads, I used media_sessions.phtml, and looked at the actual calls
per codec, and I dont think u can really get a good call without 50-70K,
also most bandwidth providers (at least here in the UK) are
asymmetrical, so even on a 128K, u could have problems.
Having said that I have done a nice call on xlite using ilbc on dial up.
Iqbal
Kofi Obiri-Yeboah wrote:
>Just to add a few more details, Greger is right to point out the quality
>inferiority of G.723 compared to those of G711 and G729. In fact, in most
VOIP
>deployments, in order to quarantee interoperability, a minimum bandwidth of
>128K is specified. However to reach the wider "lower bandwidth areas" most
>service providers are opting for G.723 which uses either 5.3 or 6.3K. At
this
>low bandwidth transmission needs, one could literally reach "dial up modem"
>equipped areas. in fact most VOIP phone hardware and software are begining
to
>specify G.723 as their default codec. Note that until the direct media
>connection phase of a VOIP vall setup, wide bandwidth is not required. Also
>note that analogue phones have a maximim bandwidth need of 3K, hence even
the
>low quality of G.723/5.3K, compared to the average analogue phone call, is
>superior
>
>
_______________________________________________
Serusers mailing list
Serusers(a)iptel.org
http://mail.iptel.org/mailman/listinfo/serusers
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I tried G723 on hardphone on dialup over mediaproxy.
On Dialup UA end up/down stream was 24k/19k
It was nice call but sometimes both party of both end noticed voice loss.
Thanks,
MOhammad
Original Message:
-----------------
From: Iqbal iqbal(a)gigo.co.uk
Date: Fri, 27 May 2005 15:36:07 +0100
To: kofi(a)radiocomplex.com, serusers(a)iptel.org
Subject: Re: [Serusers] G723.1 vs G729
the 5.3, 6.3K are really theoretical, i dont think they include IP
overheads, I used media_sessions.phtml, and looked at the actual calls
per codec, and I dont think u can really get a good call without 50-70K,
also most bandwidth providers (at least here in the UK) are
asymmetrical, so even on a 128K, u could have problems.
Having said that I have done a nice call on xlite using ilbc on dial up.
Iqbal
Kofi Obiri-Yeboah wrote:
>Just to add a few more details, Greger is right to point out the quality
>inferiority of G.723 compared to those of G711 and G729. In fact, in most
VOIP
>deployments, in order to quarantee interoperability, a minimum bandwidth of
>128K is specified. However to reach the wider "lower bandwidth areas" most
>service providers are opting for G.723 which uses either 5.3 or 6.3K. At
this
>low bandwidth transmission needs, one could literally reach "dial up modem"
>equipped areas. in fact most VOIP phone hardware and software are begining
to
>specify G.723 as their default codec. Note that until the direct media
>connection phase of a VOIP vall setup, wide bandwidth is not required. Also
>note that analogue phones have a maximim bandwidth need of 3K, hence even
the
>low quality of G.723/5.3K, compared to the average analogue phone call, is
>superior
>
>
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