Hello
I want to have a server that supports SIP and H.323 endpoints calling eachother, so I installed GNUGK, SER and Asterisk.
GnuGK is needed for H.323 endpoints and users Accounting, Authentication and Authorization.
SER is needed for the same 3 A's but for SIP users.
Asterisk has some good funcionalities like voicemail, call holding and PSTN connection, supports SIP and H.323, but its not a realy gatekeeper and a realy SIP server.
Now,
Who is going to accept the SIP calls, Asterisk ou SER?
Who is going to redirect calls to PSTN? Asterisk or SER?
In the H.323->SIP calls, who should the gnugk contact? Asterisk or SER?
If someone is using a similar network, please give me a little help.
Thanks
Joao Pereira
I know the Zyxel 2000 Wireless phones to be bad with G729 (the default) and WEP enabled.
Usually, asterisk will sit in the media stream. As the G729 codec needs licensing asterisk would have rejected it and defaulted to G711 a-law (probably).
As ser does not sit in the media stream, probably the phones tried to use g729 as they just directly connected the media streams.
Also, inherently the sound quality will not be as good from an 8kbps codec as a 64kbps codec...
Hope this has some pointers,
Dave
-----Original Message-----
From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Daniel-Constantin Mierla
Sent: 21 October 2004 11:25
To: Joao Pereira
Cc: serusers
Subject: Re: [Serusers] faster SER
There is nothing to improve in SER as long the communication was
established. In the case a NAT would be involved you would need either a
RTP relay (rtpproxy, media proxy) or in some cases just a STUN server
(depending on SIP clients and NAT type).
On 10/21/04 12:15, Joao Pereira wrote:
>No, there was no NAT involved, both phones were in the same LAN.
>With bad comunication I mean bad audio (with eco) and with lots of moments
>with no audio at all.
>
>
Try to force the SIP clients to use some other audio codec (if they
support more than one) -- maybe one has a broken implementation of the
codec chosen for communication.
>And in the same LAN and in the same conditions, but with Asterisk, the
>comunication was perfect.
>
>
I do not know much about asterisk, it may do some media correction or
codecs translation.
Daniel
>Is there any configuration changes that I can make to inprove SER?
>
>Joao
>
>
>
>----- Original Message -----
>From: "Daniel-Constantin Mierla"
><Daniel-Constantin.Mierla(a)fokus.fraunhofer.de>
>To: "Joao Pereira" <joao.pereira(a)fccn.pt>
>Cc: <serusers(a)lists.iptel.org>
>Sent: Thursday, October 21, 2004 11:04 AM
>Subject: Re: [Serusers] faster SER
>
>
>what you mean by communication was bad? The audio quality? The signaling?
>
>SER doesn't deal with audio stream at all, just with the signaling part.
>Is there any NAT involved?
>
>Daniel
>
>On 10/21/04 11:54, Joao Pereira wrote:
>
>
>
>>Hi
>>I tried two SIP Phones Zyxel Prestige 2000 W with SER and the
>>comunication was very bad, even in a second test with SIP software
>>client and a Zyxel phone, the comunication wasnt good, then I tried
>>the same phones and software with the Asterisk server and the
>>comunication was perfect, can I configure SER to have a better
>>performance in the comunication?
>>Does anyone experienced a problem like this?
>>
>>João Pereira
>>
>>------------------------------------------------------------------------
>>
>>_______________________________________________
>>Serusers mailing list
>>serusers(a)lists.iptel.org
>>http://lists.iptel.org/mailman/listinfo/serusers
>>
>>
>>
>>
>
>
>
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Hi
Sorry if this is already answered, but i've search the archives and
couldn't find what I'm looking for.
Basically I would like to have the following scenario:
UA1 asym --> NAT/Firewall --> Ser/mediaproxy --> PSTN Gateway
PSTN Gateway is a Quintum box, we already have several of this, and it
can only support asymmetric rtp, that's why we are looking at mediaproxy
I develop the UA1 myself, and can support both symmetric or asymmetric
rtp.
My question, is this possible with Ser/mediaproxy? If it is, could
somebody give me pointer on how to configure Ser with mediaproxy
Currenty I have the call goes to the pstn gateway but no voice on each
side.
Another question, how do you know if the sip signalling is asymmetric or
not?
Thanks,
Fikri
here is my ser config:
# ----------- global configuration parameters ------------------------
debug=4 # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=yes # (cmd line: -E)
/* Uncomment these lines to enter debugging mode
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"
loadmodule "/usr/local/lib/ser/modules/textops.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/auth_db.so"
# !! Nathelper
#loadmodule "/usr/local/lib/ser/modules/nathelper.so"
# !! Mediaproxy
loadmodule "/usr/local/lib/ser/modules/domain.so"
loadmodule "/usr/local/lib/ser/modules/mediaproxy.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)
# -- 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)
# !! Nathelper
#modparam("registrar", "nat_flag", 6)
#modparam("nathelper", "natping_interval", 30) # Ping interval 30 s
#modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT
# !! Mediaproxy
modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock")
modparam("mediaproxy", "sip_asymmetrics", "/usr/local/etc/ser/sip-asymmetrics-clients")
modparam("mediaproxy", "rtp_asymmetrics", "/usr/local/etc/ser/rtp-asymmetrics-clients")
modparam("mediaproxy", "natping_interval", 60)
modparam("registrar", "nat_flag", 2)
# ------------------------- request routing logic -------------------
# main routing logic
route{
if (!mf_process_maxfwd_header("10")) {
if (method!="ACK") {
sl_send_reply("483", "Too many hops");
};
break;
};
if (msg:len >= max_len) {
if (method!="ACK") {
sl_send_reply("513", "Message too big");
};
break;
};
if (method=="REGISTER") {
if (is_from_local()) {
# Mark as NAT'ed
if (client_nat_test("3")) {
setflag(2);
force_rport();
fix_contact();
};
# if (!www_authorize("", "subscriber")) {
# www_challenge("", "0");
# break;
# } else if (!check_to()) {
# sl_send_reply("403", "Username!=To not allowed");
# break;
# };
if (!save("location")) {
sl_reply_error();
};
} else {
sl_send_reply("403", "This domain is not served here");
};
break;
};
if (method=="INVITE") {
if (!(is_from_local() || is_uri_host_local())) {
sl_send_reply("403", "Relaying is forbidden");
break;
};
t_on_failure("1");
} else if (method == "BYE" || method == "CANCEL") {
end_media_session();
};
if (loose_route()) {
if (method=="INVITE" || method=="ACK") {
use_media_proxy();
};
# end media session for BYE and CANCEL is done above
# before entering the loose route. no need to call it here
t_relay();
break;
};
# Force subsequent messages to pass trough this proxy
if (method == "INVITE") {
record_route();
};
if (client_nat_test("3") && !search("^Record-Route:")) {
# Mark as NAT'ed
force_rport();
fix_contact();
};
if (method=="INVITE") {
t_on_reply("1");
};
if (is_uri_host_local()) {
if (!lookup("location")) {
if (uri=~"^sip:[0-9]*@.*") {
log(1, "Calling gateway\n");
rewritehostport("202.172.45.110:5060");
forward(uri:host, uri:port);
# break;
} else {
sl_send_reply("404", "User not found");
break;
};
};
};
if (method=="INVITE" || method=="ACK") {
use_media_proxy();
};
if (!t_relay()) {
if (method=="INVITE" || method=="ACK") {
end_media_session();
};
sl_reply_error();
};
}
failure_route[1] {
end_media_session();
}
onreply_route[1] {
if (status=~"(183)|(2[0-9][0-9])") {
if (client_nat_test("1")) {
fix_contact();
};
use_media_proxy();
};
}
In this same topic.
Assuming that i want to use exec_dset() and exec_msg() commands to acomplish
this task.
Is possible to get the variable SIP_ORUI for example... pass it to an
external program.. the program modifies the value... then i use the SIP_ORUI
variable in my ser.cfg to change the number?
Is this possible?
Thanks in advance.
-----Mensaje original-----
De: Ricardo Martinez [mailto:rmartinez@redvoiss.net]
Enviado el: Jueves, 21 de Octubre de 2004 15:44
Para: 'serusers(a)lists.iptel.org'
Asunto: [Serusers] Changing Dialed Number.
Hello List.
I have succesfully installed SER on my Linux platform. I'm
authenticating users and authorizing calls through the RADIUS SER support.
What i want to do know is change the dialed number by a user. For this task
i would like to know if i can use the same RADIUS system for authorize a
call. For example, i make a call, the Radius message goes to my Radius
Server, the Radius server adds a parameter called Re-Direct-Number=1234567,
this parameter is taken by the SER and inserted in the R-URI parameter in
the SIP message?. Is possible to do this? If is not possible, can i use
the exec() command to do something like that?
I would appreciate any help.
Thanks in advance
Best Regards
Ricardo Martinez
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serusers(a)lists.iptel.org
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Check if you do not have SDP in your ACK. Your gateway might not
understand it.
Adrian
> Hi..i have a problem with my grandstream phones..When I try to make a
> call to a PSTN gateway the call disconnects after 15-20 seconds.But
> when I make a call between two grandstreams or between an ata 186 and
> a grandstream it is ok.I used ngrep to capture packets and it seems
> to be something about how the ACK is transmited between the
> grandstream and the gateway.The ACK is ok in the other cases.What
> seems to be the problem ?.
>
>
> Thanks
>
> _______________________________________________
> Serusers mailing list
> Serusers at iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
>
Hello List.
I have succesfully installed SER on my Linux platform. I'm
authenticating users and authorizing calls through the RADIUS SER support.
What i want to do know is change the dialed number by a user. For this task
i would like to know if i can use the same RADIUS system for authorize a
call. For example, i make a call, the Radius message goes to my Radius
Server, the Radius server adds a parameter called Re-Direct-Number=1234567,
this parameter is taken by the SER and inserted in the R-URI parameter in
the SIP message?. Is possible to do this? If is not possible, can i use
the exec() command to do something like that?
I would appreciate any help.
Thanks in advance
Best Regards
Ricardo Martinez
Hi All.
I'm using ser-0.8.99-dev10 with MySQL and when I start ser it does not load the
location table rows are not being read. This forces all UAs to reregister every
time I restart the ser proxy.
Here is a snippet from /var/log/message which shows ser complaining about the
location records being expired, but this should not happen because all the
location entries were created just moments before I restarted ser.
Can anyone explain why ser will not honor the rows in my MySQL locations table?
Regards,
Paul
Oct 21 09:42:53 sip01 /usr/local/sbin/ser[11821]: rtpp_test: RTP proxy found,
support for it enabled
Oct 21 09:43:03 sip01 /usr/local/sbin/ser[11827]: Binding
'1444(a)mycompany.com','sip:1444@11.22.33.44:50040;user=phone' has expired
Oct 21 09:43:03 sip01 /usr/local/sbin/ser[11827]: Binding
'1011(a)mycompany.com','sip:1011@11.22.33.44:50009;user=phone' has expired
[root@sip01 root]#
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Hi..i have a problem with my grandstream phones..When I try to make a call
to a PSTN gateway the call disconnects after 15-20 seconds.But when I make
a call between two grandstreams or between an ata 186 and a grandstream it
is ok.I used ngrep to capture packets and it seems to be something about
how the ACK is transmited between the grandstream and the gateway.The ACK
is ok in the other cases.What seems to be the problem ?.
Thanks
This time its about a problem I'm having with serweb. I think I've set
up everything correctly in config.php. But, I'm getting the following
error when I point my browser at serweb/admin:
Fatal error: Cannot redeclare class csub_not in
/var/www/serweb/config.php on line 6
Host information:
Debian unstable
Apache 1.3.31-6
MySQL 4.0.21-7
php4 4.3.9-1
php-pear-log 1.6.0-1
There are several php-mysql apps working on this box, such as
phpmyadmin. register_globals is on. I'm really stuck on this. help!
johno
Hello
I want to have a server that supports SIP and H.323 endpoints calling eachother, so I installed GNUGK, SER and Asterisk.
GnuGK is needed for H.323 endpoints and users Accounting, Authentication and Authorization.
SER is needed for the same 3 A's but for SIP users.
Asterisk has some good funcionalities like voicemail, call holding and PSTN connection, supports SIP and H.323, but its not a realy gatekeeper and a realy SIP server.
Now,
Who is going to accept the SIP calls, Asterisk ou SER?
Who is going to redirect calls to PSTN? Asterisk or SER?
In the H.323->SIP calls, who should the gnugk contact? Asterisk or SER?
If someone is using a similar network, please give me a little help.
Thanks
Joao Pereira
