27 Dec
2006
27 Dec
'06
2:08 p.m.
Hi!
The problem is that all rewriting functions (subst_body, replace,
use_media_proxy ...) work on the original received text message. This
causes problem if 2 functions rewrite the same part of the SIP message.
A bad hack would be to loop the message to openser again before sending
it to the B2BUA.
client ---1.---> openser ---3.---> B2Bua
^ \
| \
|--2.-|
The first time the request is processed oyu use use_media_proxy, the
second time the request is processd you use subst_body.
But why doing it at the SIP proxy? Isn't is possible to change codec
preference on the Asterisk?
regards
klaus
Oleg A Chernov wrote:
Hello everyone!
My setup is as follow:
for local PSTN calls
UA (with G.711 and G.729) - my Openser - my Asterisk B2BUA for prepaid -
PSTN gateway (G.711 codec only)
for international calls
UA (with G.711 and G.729) - my Openser - my Asterisk B2BUA for prepaid -
ITSP gateway (G.729 codec only)
B2BUA on the second leg (B2BUA - gateway) always negotiate the right codec, so
no problem, but on the first leg (UA - B2BUA) it always negotiate codec that
is more prefered by the client UA side. So we can get situation, when the
codecs are different on the both legs.
Inspired by the
http://lists.iptel.org/pipermail/serusers/2006-August/030129.html
I decided to normalize codec preference.
Sorry for such a long introduction, at last here goes my problem.
I have the following code in my openser.cfg in INVITE section
# INVITE section
route[3] {
...
if ( route to G.711 capable gateway) {
subst_body("/^m=audio(.*)RTP\/AVP(.*) 8 (.*)/m=audio\1RTP\/AVP 8 \2 \3/ig");
....
}
...
}
and it works like a charm for non-nated clients with real IPs. But when we
should fire use_media_proxy() for a nated one client we get the problem.
SDP body before our substitution (UA-Openser):
v=0
o=20000 12893 12893 IN IP4 192.168.2.134
s=ATA186 Call
c=IN IP4 192.168.2.134
t=0 0
m=audio 16384 RTP/AVP 0 18 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=yes
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
After (Openser - B2BUA):
v=0
o=20000 12893 12893 IN IP4 192.168.2.134
s=ATA186 Call
c=IN IP4 217.77.208.184
t=0 0
m=audio 16384 RTP/AVP 8 0 18 101\01560136\012a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=yes
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
So our subst_body() rearrange codecs as expected, but use_media_proxy() (it
her fault, I think) inserted the mediaproxy fixed port number 60136 between
symbols with ASCII codes \015 CR and \012 LF. This behavior was accepted by
stable Openser 1.1 and latest cvs 1.2.
So the question is :
What to do and who is guilty ? :-)
--
Klaus Darilion
nic.at