[OpenSIPS-Users] one way audio in voip clients
Iam new to opensips.i have installed successfully opensips on my pc. i have registered two voip clients. but only one way audio is working. please help me from this issue. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] one way audio problem
i configured and installed opensips successfully,i have registerd two clients:- 6005 :- 192.168.2.48 6006:- 192.168.2.50 i can hear only one way audio.iam using wifi standalone router to communicate with clients.i have searched in the blogs to slove the problem. by seeing the blogs i came to know that rtptproxy would slove problem. i have integrated opensips with rtpproxy module successfully, but still the problem is not sloved. iam attaching the sip trace file and opensips.cfg file. please help me from this issue. Thank you Nandini opensips.cfg Description: Binary data sip Description: Binary data ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] one way audio problem
Thanku very much for your prompt response,
iam new to opensips, please tell me where to add these lines in opensips.cfg
route[nat_check] {
if (client_nat_test("3")) {
force_rport();
fix_contact();
nat_keepalive();
}
}
The above lines i have added in opensips.cfg under Routing logic,
when i start opensips server,iam getting errors,
please help me.
Nandini
On Tue, May 7, 2013 at 2:30 PM, aamir chougule wrote:
> Hi Nandini,
>
> The parameters and modules that you need to turn ON in your opensips.cfg
> file:
>
> loadmodule "nat_traversal.so"
>
> The above line load the module and the given below paragraph will set to
> test the parameters.
>
> route[nat_check] {
> if (client_nat_test("3")) {
> force_rport();
> fix_contact();
> nat_keepalive();
> }
> }
>
> Everytime you route a call first test the calls through the
> route(nat_check) which will fix all the NAT handling parameters.
>
> For e.g. if you are gonna route INVITE request then you need to do it like
> this:
>
> if(is_method("INVITE")) {
> route(invite_requests);
> exit;
> }
>
> route[invite_requests] {
> route(nat_check);
>
> if(!lookup("location")) {
> sl_send_reply("404", "User Not registered");
> exit;
> }
> t_on_reply("user_reply");
> t_relay();
> exit;
> }
>
> Its just an example that how I do it and always you can explore things and
> read the modules provided by OpenSIPS and upgrade yourself to use this
> server in all possible cases.
>
> Regards,
>
> Aamir Chougule
> Cell: 09167989111
>
> --
> *From:* Aamir
>
> *To:* OpenSIPS users mailling list
> *Sent:* Tuesday, 7 May 2013 1:58 PM
> *Subject:* Re: [OpenSIPS-Users] one way audio problem
>
> Hi Nandini,
>
> You actually need to turn on the nat_traversal module I guess, will pass
> you the parameters if I get time to do so.
>
> --Aamir
> --- Sent from My BlackBerry ---
>
> -Original Message-
> From: sermj 2012
> Sender: users-boun...@lists.opensips.org
> Date: Tue, 7 May 2013 13:49:04
> To:
> Reply-To: OpenSIPS users mailling list
> Subject: [OpenSIPS-Users] one way audio problem
>
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[OpenSIPS-Users] sipclient registration with domain name.
i installed opensips,server sucessfully. in my pc i have installed Domain name server. my pc is working fine with DNS. but iam unable to register the clients with domain name. instead of i can register the clients with IP address. please help me nandini ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Error 404 not here
Hi All, My VoIP clients are registerd with opensips. When i tried to call other end client, i can hear only one way audio in clients (In caller),somewhere is going wrong and i am not getting audio (Voice) in both sides. By using tshark tracer i can see there is an '404 not here' status code. PS: I am working under standalone network infrastructure, and VoIP phones are Wi-Fi enabled. I just reinstalled my operating system (Ubuntu 10.04) and Opensips for several times,but unable to resolve this issue. what may be the reason for this '404 Not here' issue? and how can i solve it? Please help me, i have just stuck with this issue from last 15 days. Find the attachment, which is sip trace file. I hope you could help me... sip Description: Binary data ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] opensips + asterisk,
Dear all, i have integrated asterisk with opensips server with the help of these link, http://www.voip-info.org/wiki/view/Asterisk+and+OpenSER+integration my voip clients are registered well,and there is audio on both the sides in WiFi network. but the same configuration is not working under Wimax network. The clients are registered ,but when the call is initiated there is no audio. what could be the problem,iam not understanding. please help me in this issue... Thanks in advance Nandini ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] asterisk + opensips
Dear aamir, As you said i have checked the RTP packets, of incoming and outgoing call. I am working under stand-alone independent network. And i don’t think so, there is an NAT enabled in my own network. Please find the below logs ngrep based using in WiMAX network. # U 192.168.2.78:27147 -> 192.168.2.40:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.40:5060;branch= z9hG4bKac49.46354042.0;rport=5060;received=192.168.2.40..Via: SIP/2.0/UDP 192.168.2.66:33270;received=19 2.168.2.66;branch=z8hG4bKedd26b72-24535979666936-keywe54;rport=33270..Max-Forwards: 70..To: "6006";tag=188069316431..From: "6010" ;tag=398687640631..Call-ID: edb8dc55-5fd7-154f-98-df-00210705d...@192.168.2.66..cseq: 1524 INVITE..Contact: ..User-Agen t: KEYWESOFT SIP Agent v3.1..Expires: 3600..Content-Length: 247..Content-Type: application/sdpv=0..o=- 295794397 37438 IN IP4 192.168.2.78..s=KEYWESOFT SIP Ag ent v3.1..c=IN IP4 192.168.2.78..t=0 0..m=audio 20934 RTP/AVP 8 0 101..a=rtpmap:8 PCMA/8000/1..a=rtpmap:0 PCMU/8000/1..a=rtpmap:101 telephone-event/8000..a=fmtp:1 01 0-15..a=sendrecv.. # U 192.168.2.40:5060 -> 192.168.2.66:33270 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.66:33270;received=192.168.2.66;branch=z8hG4bKedd26b72-24535979666936-keywe54;rport=33270..Max-Forwards: 70..To: "6006"< sip:6006@192.168.2.40>;tag=188069316431..From: "6010" < sip:6010@192.168.2.40>;tag=398687640631..Call-ID: edb8dc55-5fd7-154f-98-df-00210705d...@192.168.2.66..cseq: 1524 INVITE..Contact: ..User-Agent: KEYWESOFT SIP Agent v3.1..Expires: 3600..Content-Length: 247..Content-Type: application/sdpv =0..o=- 295794397 37438 IN IP4 192.168.2.78..s=KEYWESOFT SIP Agent v3.1..c=IN IP4 192.168.2.78..t=0 0..m=audio 20934 RTP/AVP 8 0 101..a=rtpmap:8 PCMA/8000/1..a=rt pmap:0 PCMU/8000/1..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv.. # U 192.168.2.66:33270 -> 192.168.2.40:5060 ACK sip:6006@192.168.2.40:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.2.66:33270;branch=z8hG4bKbc73cd57-245451155943079-keywe55;rport..Max-Forwards: 70..To: "6006" ;tag=188069316431..From: "6010"< sip:6010@192.168.2.40>;tag=398687640631..Call-ID: edb8dc55-5fd7-154f-98-df-00210705d...@192.168.2.66..cse q: 1524 ACK..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..Contact: ;expires=3600..User-Agent: K EYWESOFT SIP Agent v3.1..Expires: 3600..Content-Length: 0 # U 192.168.2.40:5060 -> 192.168.2.78:27147 ACK sip:6006@192.168.2.78:27147 SIP/2.0..Record-Route: ..Via: SIP/2.0/UDP 192.168.2.40:5060 ;branch=z8hG4bKbc73cd57-245451155943079-keywe55..V ia: SIP/2.0/UDP 192.168.2.66:33270;received=192.168.2.66;branch=z8hG4bKbc73cd57-245451155943079-keywe55;rport=33270..Max-Forwards: 69..To: "6006" ;tag=188069316431..From: "6010";tag=398687640631..Call-ID: edb8dc55-5fd7-154f-98-df-00210705d...@192.168.2.66..cseq: 1524 ACK.. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..Contact: ;expires=3600..User-Agent: KEYWESOFT SIP Agent v3.1..Expires: 3600..Content-Length: 0..P-hint: usrloc applied # U 192.168.2.66:20932 -> 192.168.2.78:20934 .~.y...U.WPTU...UWWQSQ.V]RVQQ]P...UPVUUUT..UU...P]RRRP\RQWPYSQSPTQWQRQTWUUT..QQW..VU...TUT UQ # U 192.168.2.66:20932 -> 192.168.2.78:20934 .~.yPQVP]R[X\QVQ^Q..UR]R\\_SWVTQVS^RPQSRWPWWWVT.T.SRSPS\QVPTVUTPU...V]PQR_XYQP^]\^\..UVUWVUWRXZXSQT.VT..PR]ZFE YRSR\R]^R] # U 192.168.2.66:20932 -> 192.168.2.78:20934 ...@.~.yQ..WRS_YZZ_U.UTP]X]PPQWSQQVVVY[SWQ]PWVVVQR\PTU..QPVW...U.WVWQWR]QUWS_QT.UU.UTPSU...WU.UPQWW... V\ # U 192.168.2.66:20932 -> 192.168.2.78:20934 .~.y_PWTTVPU.]]V]X^PTPSV.VY\]]VTUQT.RZYSWQR_XXDGAZ_PU...UT...VPSQR_]RRYYQ..WWW VPQWQRRT.. # U 192.168.2.66:20932 -> 192.168.2.78:20934 .~.y.TVSY[ZZ_\\SPWT...UWU.VV\[EEGX^YXZ_W..VQW.VTQR]RSV..W\]PVPWTT...QTTQSST... .. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
