[OpenSIPS-Users] FW: Audio and Video not working for different otuside netrwork
Rajesh Babu
rajesh.babu at goodcoresoft.com
Mon Sep 30 16:35:32 CEST 2013
J Thousand thanks to you mike, with this steps mentioned in the wiki was
able to connect the cal. I guess i have to tune the same for the
performance. The calls are having lot of delay, please advise me on the
same. One more time thanks a lot mike (mike at ultra.net.br) you made my day.
From: mike.tesliuk at ultra.net.br [mailto:mike.tesliuk at ultra.net.br] On Behalf
Of Mike Tesliuk
Sent: Sunday, 29 September, 2013 2:21 AM
To: rajesh.babu at goodcoresoft.com
Subject: Re: [OpenSIPS-Users] FW: Audio and Video not working for different
otuside netrwork
the same for this
Take a look over this howto
http://opensips.com.br/wiki/index.php?title=Oopensips_Nat_script_with_RTPPRO
XY_-_English
2013/9/28 Mike Tesliuk <mike at ultra.net.br>
Take a look over this howto
http://opensips.com.br/wiki/index.php?title=Oopensips_Nat_script_with_RTPPRO
XY_-_English
2013/9/28 Mike Tesliuk <mike at ultra.net.br>
---- if i say something wrong please somebody correct my ----
Hello Rajesh,
You are using the nat_uac_test with parameter 23, this means parameters 16,
4, 2, 1 , what means
. 1 - Contact header field is searched for occurrence of RFC1918
addresses.
. 2 - the "received" test is used: address in Via is compared
against source IP address of signaling
. 4 - Top Most VIA is searched for occurrence of RFC1918 addresses
. 16 - test if the source port is different from the port in Via
i dont know if you understand but this is a binary count, you can check in
this way
0010111 -> this is what you turn on
in this case, if your package does not contains an Private ip address on
contact header, or does not contains a received on VIA different from the ip
address of the signalling, does not contais on VIA an private ip address and
the source port is not different from port on VIA , so your rule will not
match (just on match is enought)
Look at this invite below (sended from a zoiper)
204.16.0.26:60340 -> 204.16.1.50:5060
INVITE sip:101 at 204.16.1.50 <mailto:sip%3A101 at 204.16.1.50> ;transport=UDP
SIP/2.0.
Via: SIP/2.0/UDP
75.74.203.73:60340;branch=z9hG4bK-d8754z-f6a3eadc786e7359-1---d8754z-;rport.
Max-Forwards: 70.
Contact: <sip:102 at 75.74.203.73:60340;transport=UDP>.
To: <sip:101 at 204.16.1.50 <mailto:sip%3A101 at 204.16.1.50> ;transport=UDP>.
From: "102"<sip:102 at 204.16.1.50 <mailto:sip%3A102 at 204.16.1.50>
;transport=UDP>;tag=489f8f45.
Call-ID: ZGNhYTQzNjIyOGFkYWNhOWQ3ZmQ2ZDVkYjhiNGI4MGE..
CSeq: 1 INVITE.
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
SUBSCRIBE.
Content-Type: application/sdp.
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri.
User-Agent: Zoiper Communicator 2.04.10164 rev.10204.
Allow-Events: presence, kpml.
Content-Length: 352.
you can see the ip on signalling coming from 204.16.0.26 port 60340
on via you have 75.74.203.73:60340, so you have a different ip address from
signalling or via , in this case you will set the NAT variable, but check
the invite below.
#
U 204.16.0.26:5062 -> 204.16.1.50:5060
INVITE sip:102 at 204.16.1.50 <mailto:sip%3A102 at 204.16.1.50> SIP/2.0.
Via: SIP/2.0/UDP 204.16.0.26:5062;branch=z9hG4bK1527256431.
From: "Mike" <sip:101 at 204.16.1.50 <mailto:sip%3A101 at 204.16.1.50>
>;tag=1050377705.
To: <sip:102 at 204.16.1.50 <mailto:sip%3A102 at 204.16.1.50> >.
Call-ID: 83821284 at 10.254.254.6.
CSeq: 1 INVITE.
Contact: <sip:101 at 204.16.0.26:5062>.
Content-Type: application/sdp.
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER,
SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE.
Max-Forwards: 70.
User-Agent: Yealink SIP-T20P 9.70.0.121.
Supported: replaces.
Allow-Events: talk,hold,conference,refer,check-sync.
Content-Length: 304
You have the same port on signalling and on VIA, in this case the rule will
no match and variable will not be set and this is a phone behind a nat
so, you should try to remove the if where you call the rtpproxy offer and
answer (just for test purpose)
you should increment you debug info too
/* uncomment the following lines to enable debugging */
#debug=6
#fork=no
#log_stderror=yes
2013/9/28 Rajesh Babu <rajesh.babu at goodcoresoft.com>
Hi,
I have attached the logs and my routing file @
http://pastebin.com/hu0bQGVw
Please help me out in nailing this.
Thanks
Rajesh
From: users-bounces at lists.opensips.org
[mailto:users-bounces at lists.opensips.org] On Behalf Of Mike Tesliuk
Sent: Friday, 27 September, 2013 11:25 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] FW: Audio and Video not working for different
otuside netrwork
If possible, paste your route file too
2013/9/27 Mike Tesliuk <mike at ultra.net.br>
start your opensips in debug mode, try to make the call, get all the message
and paste in some pastebin website and show us the link
2013/9/27 Rajesh Babu <rajesh.babu at goodcoresoft.com>
I am getting Error 483, too many Hops, There is no other error messages i am
getting. Please some one help me out in this
From: users-bounces at lists.opensips.org
[mailto:users-bounces at lists.opensips.org] On Behalf Of Rajesh Babu
Sent: Friday, 27 September, 2013 6:08 PM
To: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] FW: Audio and Video not working for different
otuside netrwork
HI Mike,
Now the RTP is up and i am getting this message on my logs
[root at centos64 rtpproxy-1.2.0]# tailf /var/log/messages
Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17800]:
INFO:core:probe_max_sock_buff: using rcv buffer of 448 kb
Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17810]:
INFO:rtpproxy:rtpp_test: rtp proxy <udp:10.10.10.123:7890> found, support
for it enabled
Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17812]:
INFO:rtpproxy:rtpp_test: rtp proxy <udp:10.10.10.123:7890> found, support
for it enabled
Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17815]:
INFO:rtpproxy:rtpp_test: rtp proxy <udp:10.10.10.123:7890> found, support
for it enabled
Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17808]:
INFO:rtpproxy:rtpp_test: rtp proxy <udp:10.10.10.123:7890> found, support
for it enabled
Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17811]:
INFO:rtpproxy:rtpp_test: rtp proxy <udp:10.10.10.123:7890> found, support
for it enabled
Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17809]:
INFO:rtpproxy:rtpp_test: rtp proxy <udp:10.10.10.123:7890> found, support
for it enabled
Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17814]:
INFO:rtpproxy:rtpp_test: rtp proxy <udp:10.10.10.123:7890> found, support
for it enabled
Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17809]:
WARNING:drouting:dr_load_routing_info: table "dr_rules" is empty
Sep 28 01:59:43 centos64 opensips: INFO:core:daemonize: pre-daemon process
exiting with 0
But my test tool is not connecting back my server. Is there any mistake i am
doing.
Thanks
Rajesh
From: users-bounces at lists.opensips.org
[mailto:users-bounces at lists.opensips.org] On Behalf Of Rajesh Babu
Sent: Friday, 27 September, 2013 2:34 PM
To: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] FW: Audio and Video not working for different
otuside netrwork
Hi Mike,
This is log i am geting wheni try to start the service
[root at centos64 rtpproxy-1.2.0]# tailf /var/log/messages
Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12839]:
WARNING:rtpproxy:rtpp_test: support for RTP proxy <udp:localhost:7890> has
been disabled temporarily
Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12833]:
ERROR:rtpproxy:send_rtpp_command: can't send command to a RTP proxy
Connection refused
Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12833]:
ERROR:rtpproxy:send_rtpp_command: proxy <udp:localhost:7890> does not
respond, disable it
Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12833]:
WARNING:rtpproxy:rtpp_test: can't get version of the RTP proxy
Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12833]:
WARNING:rtpproxy:rtpp_test: support for RTP proxy <udp:localhost:7890> has
been disabled temporarily
Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12832]:
ERROR:rtpproxy:send_rtpp_command: can't send command to a RTP proxy
Connection refused
Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12832]:
ERROR:rtpproxy:send_rtpp_command: proxy <udp:localhost:7890> does not
respond, disable it
Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12832]:
WARNING:rtpproxy:rtpp_test: can't get version of the RTP proxy
Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12832]:
WARNING:rtpproxy:rtpp_test: support for RTP proxy <udp:localhost:7890> has
been disabled temporarily
Sep 27 22:29:03 centos64 opensips: INFO:core:daemonize: pre-daemon process
exiting with 0
From: users-bounces at lists.opensips.org
[mailto:users-bounces at lists.opensips.org] On Behalf Of Mike Tesliuk
Sent: Thursday, 26 September, 2013 10:25 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] FW: Audio and Video not working for different
otuside netrwork
When you use the residential script almost all configuration come alredy
working for this
i have a tutorial (in portuguese ( i think that i should translate to
english :) )) , where you can see a routing script working with nat
http://opensips.com.br/wiki/index.php?title=Opensips_1.9
You can take a look at modules documentation too
http://www.opensips.org/html/docs/modules/1.9.x/nathelper.html
http://www.opensips.org/html/docs/modules/1.9.x/rtpproxy.html
There is on this maillist too a lot of discussions about this, below you can
see one case
http://opensips.org/pipermail/users/2011-January/016130.html
If you get some information from an old version of opensips probably will be
necessary to take a look on the module documentation to check about little
diferences , but i think that this is the start point :)
and if you is new to opensips i recommend to you the book about opensips (
http://www.packtpub.com/building-telephony-systems-with-opensips-1-6/book )
2013/9/26 Rajesh Babu <rajesh.babu at goodcoresoft.com>
Hi Mike,
Thanks for the response, I am totally new to this world, can you please
help me by directing to on how to configure links. It will be great.
Thanks in advance
Regards
Rajesh
From: users-bounces at lists.opensips.org
[mailto:users-bounces at lists.opensips.org] On Behalf Of Mike Tesliuk
Sent: Thursday, 26 September, 2013 12:25 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] FW: Audio and Video not working for different
otuside netrwork
you should configure the nathelper and rtpproxy, this should help in you
issue.
2013/9/26 Rajesh Babu <rajesh.babu at goodcoresoft.com>
Hi,
I am new to the OpenSIP world. I have installed a OpenSIP on my network.
If i make a Call inside the network between two users i don't have any
issue, where as from outside the network, even though i can see the user
registered in my server i am not able to call registered user (I see the
user in my UL show listing). The call is established but i am not able to
talk (Mean the audio and video are not getting transffered).
Where as messages are going fine without any issue. I guess it is because
message transmit over XMPP where calls on SIP right.
I am really struck and i don't know how to proceed, please help me out
Thanks
Rajesh
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