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

Sat Sep 28 19:37:09 CEST 2013

Take a look over this howto

http://opensips.com.br/wiki/index.php?title=Oopensips_Nat_script_with_RTPPROXY_-_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;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****
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users****
>>
>>  ****
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users****
>>
>>  ****
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users****
>>
>> ** **
>>
>> ** **
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.opensips.org/pipermail/users/attachments/20130928/12753833/attachment-0001.htm>