9 Jul
2007
9 Jul
'07
4:50 p.m.
Hi, I'm starting using OpenSer with Asterisk.
In calls passing through Asterisk the caller or callee get music on hold of
Asterisk when is put on hold. But of course, direct calls between OpenSer
users don't get this music.
I'm thinking in the posibility of detecting "on hold" reinvites (maybe
reading
the "a=sendonly" of SDP body) and rewriting the SDP IP contact to a music on
hold server.
I have no idea if this is possible. The first issue is how to set up a RTP
music streamer for pointing there the SDP c header. Does "this" RTP music
streamer exist? of course Asterisk is not a solution since I don't want to
manage INVITE's, just rewrite the SDP.
Second issue is: could be rewriting the SDP always sucessfull? what about
cases with more than one RTP session as audio+video?
So, is there any solution for offering music on hold with OpenSer?
Thanks a lot.
--
Iñaki Baz Castillo
ibc(a)in.ilimit.es
10 Jul
10 Jul
7:33 p.m.
Im also interested in getting MOH working with just OpenSER.
I was looking at this a while ago and I couldn't see a nice way to do it,
Phone A has to basically setup a call between Phone B and your MOH server.
Doing this without some sort of custom module would be near impossible (if I get time ill
start work on a module but im really not very good!).
If anyone else has tackled this problem or has any info I'd be happy to hear it.
Thanks.
-----Original Message-----
From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On Behalf Of Iñaki Baz
Castillo
Sent: 09 July 2007 14:51
To: users(a)openser.org
Subject: [OpenSER-Users] Rewrite SDP for music of hold
Hi, I'm starting using OpenSer with Asterisk.
In calls passing through Asterisk the caller or callee get music on hold of
Asterisk when is put on hold. But of course, direct calls between OpenSer
users don't get this music.
I'm thinking in the posibility of detecting "on hold" reinvites (maybe
reading
the "a=sendonly" of SDP body) and rewriting the SDP IP contact to a music on
hold server.
I have no idea if this is possible. The first issue is how to set up a RTP
music streamer for pointing there the SDP c header. Does "this" RTP music
streamer exist? of course Asterisk is not a solution since I don't want to
manage INVITE's, just rewrite the SDP.
Second issue is: could be rewriting the SDP always sucessfull? what about
cases with more than one RTP session as audio+video?
So, is there any solution for offering music on hold with OpenSer?
Thanks a lot.
--
Iñaki Baz Castillo
ibc(a)in.ilimit.es
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9:03 p.m.
Iñaki Baz Castillo wrote:
Hi, I'm starting using OpenSer with Asterisk.
In calls passing through Asterisk the caller or callee get music on hold of
Asterisk when is put on hold. But of course, direct calls between OpenSer
users don't get this music.
I'm thinking in the posibility of detecting "on hold" reinvites (maybe
reading
the "a=sendonly" of SDP body) and rewriting the SDP IP contact to a music on
hold server.
You would also have to tell the music server where to send the rtp
stream. And you'd have to keep dialog state on the proxy in order to
capture the off-hold INVITE and tell the music server to stop sending media.
This all leads to a sip proxy acting as a B2BUA/3PCC server which needs
a lot of resources and is difficult to implement. I don't say it is
impossible but adding this functionality to openser would be quite a task.
Unfortunately even the IETF has no clear message on music on hold, no
wonder sip phones normally do not support it in a standards compliant
way. There is a draft called "SIP Service Examples" that already has 12
revisions and that includes an example on how music on hold should be
implemented.
http://www1.tools.ietf.org/html/draft-ietf-sipping-service-examples-12#sect…
Previous revisions of this draft followed a 3PCC model for music on hold
(http://tech-invite.com/Ti-sip-service-3.html), whereas revision 12 now
implements music on hold with a call transfer.
One can only hope that we will see products implementing these procedures.
Pingtel seems to offer a music-on-hold server for the 3PCC model:
http://interop.pingtel.com/#moh
And SNOM phones seem to work fine together with the pingtel/sipX
solution:
http://sipx-wiki.calivia.com/index.php/HowTo_configure_SNOM_SIP_phone_with_…
Christian
I have no idea if this is possible. The first issue is
how to set up a RTP
music streamer for pointing there the SDP c header. Does "this" RTP music
streamer exist? of course Asterisk is not a solution since I don't want to
manage INVITE's, just rewrite the SDP.
Second issue is: could be rewriting the SDP always sucessfull? what about
cases with more than one RTP session as audio+video?
So, is there any solution for offering music on hold with OpenSer?
Thanks a lot.
11 Jul
11 Jul
12:07 p.m.
El Tuesday 10 July 2007 20:03:53 Christian Schlatter escribió:
Iñaki Baz Castillo wrote:
Thanks a lot. I see that it's not an easy task. I'll try it whena have some
tiem and if I get something decent I''ll tell it here.
Regards.
--
Iñaki Baz Castillo
ibc(a)in.ilimit.es
Hi, I'm starting using OpenSer with
Asterisk.
In calls passing through Asterisk the caller or callee get music on hold
of Asterisk when is put on hold. But of course, direct calls between
OpenSer users don't get this music.
I'm thinking in the posibility of detecting "on hold" reinvites (maybe
reading the "a=sendonly" of SDP body) and rewriting the SDP IP contact to
a music on hold server.
You would also have to tell the music server where to send the rtp
stream. And you'd have to keep dialog state on the proxy in order to
capture the off-hold INVITE and tell the music server to stop sending
media.
This all leads to a sip proxy acting as a B2BUA/3PCC server which needs
a lot of resources and is difficult to implement. I don't say it is
impossible but adding this functionality to openser would be quite a task.
Unfortunately even the IETF has no clear message on music on hold, no
wonder sip phones normally do not support it in a standards compliant
way. There is a draft called "SIP Service Examples" that already has 12
revisions and that includes an example on how music on hold should be
implemented.
http://www1.tools.ietf.org/html/draft-ietf-sipping-service-examples-12#sect
ion-2.3
Previous revisions of this draft followed a 3PCC model for music on hold
(http://tech-invite.com/Ti-sip-service-3.html), whereas revision 12 now
implements music on hold with a call transfer.
One can only hope that we will see products implementing these procedures.
Pingtel seems to offer a music-on-hold server for the 3PCC model:
http://interop.pingtel.com/#moh
And SNOM phones seem to work fine together with the pingtel/sipX
solution:
http://sipx-wiki.calivia.com/index.php/HowTo_configure_SNOM_SIP_phone_with_
sipX#Music_On_Hold_.28MoH.29 
8:06 p.m.
New subject: [OpenSER-Users] Radius - prevent start record for ACK ?
I am using freeradius with openser 1.0.1.
My problem is that openser is emitting TWO radius start records: One for the
INVITE with SIP method = 1 (INVITE) and also a second start record for SIP
Method = 4 (ACK)
Is is normal that a start record is also emitted for the ACK ?
As long as the INVITE record is sent before the ACK record that is no real
problem.
However every-now-and-then (no pattern observed) Openser sends the Radius
start record with SIP Method = 4 (ACK) before the Record with SIP Method = 1
for the INVITE.
The problem is that the Radius record for the ACK does not contain the
username from the credentials so that the username is not correctly set in
the Radius record, instead of the correct username the callerid is used as
the username.
How do I prevent that openser sends start records for the ACK ? Quite
frankly I do not understand why a start record for ACK is emitted since I
set the radius_flag only for INVITEs.
the relevant parts from my configuration:
modparam("acc", "radius_config",
"/usr/local/etc/radiusclient-ng/radiusclient.conf")
modparam("acc", "radius_flag", 1)
modparam("acc", "log_flag", 1)
modparam("acc", "service_type", 15)
route{
.....
# If this is an INVITE and not RE-INVITE start accounting
if(is_method("INVITE") && !has_totag()){
# set the acc flags
setflag(1);
};
if (is_method("ACK")) { #hop-by-hop ACKs are absorbed here
t_newtran();
# drop broken ACKs
exit;
};
Any hints are welcome.
TIA
Gerry
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12 Jul
12 Jul
9:22 a.m.
New subject: [OpenSER-Users] Radius - prevent start record for ACK ?
Maybe you have set report_ack ?
http://www.openser.org/docs/modules/1.2.x/acc.html#AEN217
klaus
G.Jacobsen wrote:
I am using freeradius with openser 1.0.1.
My problem is that openser is emitting TWO radius start records: One for the
INVITE with SIP method = 1 (INVITE) and also a second start record for SIP
Method = 4 (ACK)
Is is normal that a start record is also emitted for the ACK ?
As long as the INVITE record is sent before the ACK record that is no real
problem.
However every-now-and-then (no pattern observed) Openser sends the Radius
start record with SIP Method = 4 (ACK) before the Record with SIP Method = 1
for the INVITE.
The problem is that the Radius record for the ACK does not contain the
username from the credentials so that the username is not correctly set in
the Radius record, instead of the correct username the callerid is used as
the username.
How do I prevent that openser sends start records for the ACK ? Quite
frankly I do not understand why a start record for ACK is emitted since I
set the radius_flag only for INVITEs.
the relevant parts from my configuration:
modparam("acc", "radius_config",
"/usr/local/etc/radiusclient-ng/radiusclient.conf")
modparam("acc", "radius_flag", 1)
modparam("acc", "log_flag", 1)
modparam("acc", "service_type", 15)
route{
.....
# If this is an INVITE and not RE-INVITE start accounting
if(is_method("INVITE") && !has_totag()){
# set the acc flags
setflag(1);
};
if (is_method("ACK")) { #hop-by-hop ACKs are absorbed here
t_newtran();
# drop broken ACKs
exit;
};
Any hints are welcome.
TIA
Gerry
___________________________________________________________
All new Yahoo! Mail "The new Interface is stunning in its simplicity and ease of
use." - PC Magazine
http://uk.docs.yahoo.com/nowyoucan.html
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5 comments
5 participants
participants (5)
-
Christian Schlatter -
G.Jacobsen -
Iñaki Baz Castillo -
Klaus Darilion -
Morgan Gilroy