[Kamailio-Users] I don't have asterisk audio to openser - mediaproxy

Wed Oct 29 07:10:12 CET 2008

Hi list is making tests with openser 1.3.2 and mediaproxy to solve the nat, I have gotten myself an ip it public with my supplier, I have two network cards in the pc that I am using for openser and mediaproxy together with asterisk, making tests with mediaproxy 1.9.1 when I receive a call from the pstn through asterisk I don't have audio, if I call to the pstn they listen to me well .

From: "Ventas" <sip:112 at 192.168.10.1>;tag=69451218021829df 
To: <sip:2685249 at 192.168.10.1>;tag=329cfeaa6ded039da25ff8cbb8668bd2.b1b2 
Contact: <sip:112 at 192.168.10.30:5060;transport=udp> 
Supported: path 
Call-ID: fb5f5dac83056f72 at 192.168.10.30 
CSeq: 7492 ACK 
User-Agent: Grandstream GXP2020 1.1.6.16 
Max-Forwards: 70 
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE 
Content-Length: 0 

#
U +0.022110 192.168.10.30:5060 -> 192.168.10.1:5060
INVITE sip:2685249 at 192.168.10.1 SIP/2.0 
Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bKf428b928c25dad03 
From: "Ventas" <sip:112 at 192.168.10.1>;tag=69451218021829df 
To: <sip:2685249 at 192.168.10.1> 
Contact: <sip:112 at 192.168.10.30:5060;transport=udp> 
Supported: replaces, timer, path 
Proxy-Authorization: Digest username="112", realm="192.168.10.1", algorithm=MD5, uri="sip:2685249 at 192.168.10.1", nonce="4907ac8cb6dc757eb6ba5522e0fdb9786b4c3d6e", response="c40a9387fdf5de29115c1edadc7f79db" 
Call-ID: fb5f5dac83056f72 at 192.168.10.30 
CSeq: 7493 INVITE 
User-Agent: Grandstream GXP2020 1.1.6.16 
Max-Forwards: 70 
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE 
Content-Type: application/sdp 
Content-Length: 358 

v=0 
o=112 8000 8001 IN IP4 192.168.10.30 
s=SIP Call 
c=IN IP4 192.168.10.30 
t=0 0 
m=audio 5004 RTP/AVP 0 18 3 97 2 9 101 
a=sendrecv 
a=rtpmap:0 PCMU/8000 
a=rtpmap:18 G729/8000 
a=rtpmap:3 GSM/8000 
a=rtpmap:97 iLBC/8000 
a=fmtp:97 mode=20 
a=rtpmap:2 G726-32/8000 
a=rtpmap:9 G722/16000 
a=ptime:20 
a=rtpmap:101 telephone-event/8000 
a=fmtp:101 0-11 

#
U +0.003938 192.168.10.1:5060 -> 192.168.10.30:5060
SIP/2.0 100 Giving a try 
Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bKf428b928c25dad03;rport=5060 
From: "Ventas" <sip:112 at 192.168.10.1>;tag=69451218021829df 
To: <sip:2685249 at 192.168.10.1> 
Call-ID: fb5f5dac83056f72 at 192.168.10.30 
CSeq: 7493 INVITE 
Server: OpenSER (1.3.2-notls (i386/linux)) 
Content-Length: 0 

#
U +0.000115 192.168.10.1:5060 -> 192.168.10.1:5070
INVITE sip:2685249 at 192.168.10.1:5070 SIP/2.0 
Record-Route: <sip:192.168.10.1;lr=on;ftag=69451218021829df;nat=yes> 
Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK400f.b93e5c35.0 
Via: SIP/2.0/UDP 192.168.10.30:5060;rport=5060;branch=z9hG4bKf428b928c25dad03 
From: "Ventas" <sip:112 at 192.168.10.1>;tag=69451218021829df 
To: <sip:2685249 at 192.168.10.1> 
Contact: <sip:112 at 192.168.10.30:5060;transport=udp> 
Supported: replaces, timer, path 
Call-ID: fb5f5dac83056f72 at 192.168.10.30 
CSeq: 7493 INVITE 
User-Agent: Grandstream GXP2020 1.1.6.16 
Max-Forwards: 69 
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE 
Content-Type: application/sdp 
Content-Length: 358 
P-hint: route(3)|setflag7,forcerport,fix_contact 
P-hint: inbound->inbound  
P-hint: Route[6]: mediaproxy  

v=0 
o=112 8000 8001 IN IP4 192.168.10.30 
s=SIP Call 
c=IN IP4 192.168.1.64 
t=0 0 
m=audio 35040 RTP/AVP 0 18 3 97 2 9 101 
a=sendrecv 
a=rtpmap:0 PCMU/8000 
a=rtpmap:18 G729/8000 
a=rtpmap:3 GSM/8000 
a=rtpmap:97 iLBC/8000 
a=fmtp:97 mode=20 
a=rtpmap:2 G726-32/8000 
a=rtpmap:9 G722/16000 
a=ptime:20 
a=rtpmap:101 telephone-event/8000 
a=fmtp:101 0-11 

#
U +0.000471 192.168.10.1:5070 -> 192.168.10.1:5060
SIP/2.0 100 Trying 
Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK400f.b93e5c35.0;received=192.168.10.1 
Via: SIP/2.0/UDP 192.168.10.30:5060;rport=5060;branch=z9hG4bKf428b928c25dad03 
Record-Route: <sip:192.168.10.1;lr=on;ftag=69451218021829df;nat=yes> 
From: "Ventas" <sip:112 at 192.168.10.1>;tag=69451218021829df 
To: <sip:2685249 at 192.168.10.1> 
Call-ID: fb5f5dac83056f72 at 192.168.10.30 
CSeq: 7493 INVITE 
User-Agent: Asterisk PBX 
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY 
Supported: replaces 
Contact: <sip:2685249 at 192.168.10.1:5070> 
Content-Length: 0 

I don't have a lot of experience with mediaproxy, and I have some doubts that such you see they can help me to clarify, inside the file mediaproxy.ini some options appear which I have configured them but I am not sure if it is the best way.

my scenario is the following one:

<->  UAC<-> NAT  <-> ADSL  <-> Internet <->   
eth0 wan (public ip x.x.x.x ) <- openser/mediaproxy/asterisk -> eth1 lan (192.168.11.1) <-> UAC 

[MediaProxy]

start = yes
socket = /var/run/mediaproxy.sock
group = openser
listen = None
allow = None
proxyIP = x.x.x.x (public ip)
;portRange = 60000:65000
portRange = 35000:65000
TOS = 0xb8
idleTimeout = 60
holdTimeout = 3600
forceClose = 0

[Accounting]
; one of none, radius or database
accounting = none

[Database]
user = dbuser
password = dbpass
host = dbhost
database = radius
table = radacct

[Radius]
secret = secret
server = localhost
authport = 1812
acctport = 1813
dictionaries = /etc/radiusclient-ng/dictionary, /etc/openser/radius/dictionary, /usr/share/mediaproxy/dictionary
retries = 2
timeout = 3

this couple of you line inside the openser, I don't still understand them according to the guide of ser getting started they are for asymmetric clients, but I don't find an example

modparam("mediaproxy","sip_asymmetrics","/etc/openser/sip-clients")
modparam("mediaproxy","rtp_asymmetrics","/ect/openser/rtp-clients")

somebody that can give me a good help...

regards 

rickygm

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20081028/8db13aa9/attachment.htm>