[OpenSIPS-Users] FW: Audio and Video not working for different otuside netrwork

Mike Tesliuk mike at ultra.net.br
Sat Sep 28 18:37:27 CEST 2013 

---- 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;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;transport=UDP>.
From: "102"<sip:102 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 SIP/2.0.
Via: SIP/2.0/UDP 204.16.0.26:5062;branch=z9hG4bK1527256431.
From: "Mike" <sip:101 at 204.16.1.50>;tag=1050377705.
To: <sip:102 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 <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 <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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>
>  ****
>
>
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> Users mailing list
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>
>  ****
>
>
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>
> ** **
>
> ** **
>
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