[OpenSIPS-Users] MediaProxy No Audio Problems

Fri Oct 30 13:54:12 CET 2009

Hi,

I used the 'engage_media_proxy();' in the main routing. This should create a
dialog and then enable the media proxy if needed. Please correct me if I
have misunderstood this feature.

Regards,

Ross

2009/10/29 osiris123d <duane.larson at gmail.com>

>
> In your config I don't see you calling the use_media_proxy() function
> anywhere.  This is needed in order to proxy the media.
>
> Do you have the OpenSIPS mediaproxy module installed and have the
> parameters
> set up?
> Check this link out.
> http://www.opensips.org/Resources/DocsTutorials#toc12
>
> You are going to need to use use_media_proxy() a couple of places in your
> config depending on what you want to accomplish.
>
>
>
> Ross Beer-2 wrote:
> >
> > Hi,
> >
> > I am using MediaProxy to help get over some one way audio issues, however
> > it
> > appears to be causing more problems than it is fixing.
> >
> > When I make a call between two registered phones there is no audio at
> all,
> > but when I call a gateway audio passes correctly.
> >
> > Looking at the logs it indicates that it has RTP & RTCP for one phone but
> > only RTP for the other:
> >
> >
> -------------------------------------------------------------------------------------------------------------------------------------------------------------
> > *media-relay[11366]: debug: Received new SDP offer
> > media-relay[11366]: mediaproxy.mediacontrol.StreamListenerProtocol
> > starting
> > on 50060
> > media-relay[11366]: mediaproxy.mediacontrol.StreamListenerProtocol
> > starting
> > on 50061
> > media-relay[11366]: mediaproxy.mediacontrol.StreamListenerProtocol
> > starting
> > on 50062
> > media-relay[11366]: mediaproxy.mediacontrol.StreamListenerProtocol
> > starting
> > on 50063
> > media-relay[11366]: debug: Added new stream: (audio)
> > 192.168.2.200:5638(RTP: Unknown, RTCP: Unknown) <-> <SERVER IP
> > ADDRESS>:50060 <-> <SERVER IP
> > ADDRESS>:50062 <-> Unknown (RTP: Unknown, RTCP: Unknown)
> > media-relay[11366]: debug: created new session
> > NWZmMmMwMzAxNDM5YjdiYTAwMDYxYTViNTllMTczMWI.:
> > **10002*200 at mydomain.com*<10002*200 at mydomain.com>
> > * (b62884c7) --> **10001*200 at mydomain.com* <10001*200 at mydomain.com>
> > *media-relay[11366]: debug: Got traffic information for stream: (audio)
> > 192.168.2.200:5638 (RTP: Unknown, RTCP: Unknown) <-> <SERVER IP
> > ADDRESS>:50060 <-> <SERVER IP ADDRESS>:50062 <-> Unknown (RTP: <Clients
> > Router IP>:57096, RTCP: Unknown)
> > media-dispatcher[11369]: debug: Issuing "update" command to relay at
> > <SERVER
> > IP ADDRESS>
> > media-relay[11366]: debug: updating existing session
> > NWZmMmMwMzAxNDM5YjdiYTAwMDYxYTViNTllMTczMWI.:
> > **10002*200 at mydomain.com*<10002*200 at mydomain.com>
> > * (b62884c7) --> **10001*200 at mydomain.com* <10001*200 at mydomain.com>
> > *media-relay[11366]: debug: Received updated SDP answer
> > media-relay[11366]: debug: Got initial answer from callee for stream:
> > (audio) 192.168.2.200:5638 (RTP: Unknown, RTCP: Unknown) <-> <SERVER IP
> > ADDRESS>:50060 <-> <SERVER IP ADDRESS>:50062 <-> 192.168.2.10:40022(RTP:
> > <Clients Router IP>:57096, RTCP: Unknown)
> > media-relay[11366]: debug: Got traffic information for stream: (audio)
> > 192.168.2.200:5638 (RTP: Unknown, RTCP: <Clients Router IP>:55671) <->
> > <SERVER IP ADDRESS>:50060 <-> <SERVER IP ADDRESS>:50062 <->
> > 192.168.2.10:40022 (RTP: <Clients Router IP>:57096, RTCP: Unknown)
> > media-relay[11366]: debug: Got traffic information for stream: (audio)
> > 192.168.2.200:5638 (RTP: <Clients Router IP>:55670, RTCP: <Clients
> Router
> > IP>:55671) <-> <SERVER IP ADDRESS>:50060 <-> <SERVER IP ADDRESS>:50062
> <->
> > 192.168.2.10:40022 (RTP: <Clients Router IP>:57096, RTCP: Unknown)
> > media-dispatcher[11369]: debug: Issuing "remove" command to relay at
> > <SERVER
> > IP ADDRESS>
> > media-relay[11366]: debug: removing session
> > NWZmMmMwMzAxNDM5YjdiYTAwMDYxYTViNTllMTczMWI.:
> > **10002*200 at mydomain.com*<10002*200 at mydomain.com>
> > * (b62884c7) --> **10001*200 at mydomain.com* <10001*200 at mydomain.com>
> > *media-relay[11366]: (Port 50060 Closed)
> > media-relay[11366]: (Port 50061 Closed)
> > media-relay[11366]: (Port 50062 Closed)
> > media-relay[11366]: (Port 50063 Closed)
> > media-dispatcher[11369]: debug: Got statistics: {'from_tag': 'b62884c7',
> > 'dialog_id': '841:447573368', 'start_time': 1256818436.3299999,
> > 'timed_out':
> > False, 'call_id': 'NWZmMmMwMzAxNDM5YjdiYTAwMDYxYTViNTllMTczMWI.',
> > 'to_tag':
> > '7t0mkzlie0', 'streams': [{'status': 'closed', 'caller_codec': 'G711u',
> > 'post_dial_delay': 1.252835989, 'callee_codec': '1016', 'start_time': 0,
> > 'caller_bytes': 82000, 'callee_bytes': 83600, 'caller_packets': 410,
> > 'end_time': 8, 'callee_remote': '<Clients Router IP>:57096',
> > 'caller_remote': '<Clients Router IP>:55670', 'media_type': 'audio',
> > 'callee_local': '<SERVER IP ADDRESS>:50062', 'timeout_wait': 0,
> > 'caller_local': '<SERVER IP ADDRESS>:50060', 'callee_packets': 418}],
> > 'duration': 8, 'to_uri':
> > **'10001*200 at mydomain.com'*<'10001*200 at mydomain.com'>
> > *, 'from_uri': **'10002*200 at mydomain.com'* <'10002*200 at mydomain.com'>*,
>  > 'callee_ua': 'snom370/7.3.26', 'caller_ua': 'X-Lite Beta release 4.0
> Beta
> > 2
> > stamp 55091'}*
> >
> -------------------------------------------------------------------------------------------------------------------------------------------------------------
> >
> > I am using the following OpenSIP's config:
> >
> >
> >
> > # main routing logic
> >
> > route{
> >
> > # initial sanity checks -- messages with
> >
> > # max_forwards==0, or excessively long requests
> >
> > if (!mf_process_maxfwd_header("10")) {
> >
> > sl_send_reply("483","Too Many Hops");
> >
> > exit;
> >
> > };
> >
> > if (msg:len >= 2048 ) {
> >
> > sl_send_reply("513", "Message too big");
> >
> > exit;
> >
> > };
> >
> > # !! Nathelper
> >
> > # Special handling for NATed clients; first, NAT test is
> >
> > # executed: it looks for via!=received and RFC1918 addresses
> >
> > # in Contact (may fail if line-folding is used); also,
> >
> > # the received test should, if completed, should check all
> >
> > # vias for rpesence of received
> >
> > if (nat_uac_test("31"))
> >
> > {
> >
> > # Allow RR-ed requests, as these may indicate that
> >
> > # a NAT-enabled proxy takes care of it; unless it is
> >
> > # a REGISTER
> >
> > xlog("Behind a NAT\n");
> >
> > if (is_method("REGISTER"))
> >
> > {
> >
> > fix_nated_register();
> >
> > }
> >
> > fix_nated_contact();
> >
> > force_rport(); # Add rport parameter to topmost Via
> >
> > #setbflag(6); # Mark as NATed
> >
> > };
> >
> > # we record-route all messages -- to make sure that
> >
> > # subsequent messages will go through our proxy; that's
> >
> > # particularly good if upstream and downstream entities
> >
> > # use different transport protocol
> >
> > if (!is_method("REGISTER"))
> >
> > record_route();
> >
> > if(is_method("INVITE"))
> >
> > {
> >
> > fix_nated_sdp("1");
> >
> > create_dialog();
> >
> > fix_nated_sdp("8");
> >
> > engage_media_proxy();
> >
> > }
> >
> > # subsequent messages withing a dialog should take the
> >
> > # path determined by record-routing
> >
> > if (loose_route()) {
> >
> > # mark routing logic in request
> >
> > append_hf("P-hint: rr-enforced\r\n");
> >
> > route(1);
> >
> > exit;
> >
> > };
> >
> >  if (!uri==myself) {
> >
> > # mark routing logic in request
> >
> > append_hf("P-hint: outbound\r\n");
> >
> > route(1);
> >
> > exit;
> >
> > };
> >
> > # if the request is for other domain use UsrLoc
> >
> > # (in case, it does not work, use the following command
> >
> > # with proper names and addresses in it)
> >
> > if (uri==myself)
> >
> > {
> >
> > if (is_method("REGISTER"))
> >
> > {
> >
> > # Uncomment this if you want to use digest authentication
> >
> > #if (!www_authorize("siphub.org", "subscriber")) {
> >
> > # www_challenge("siphub.org", "0");
> >
> > # return;
> >
> > #};
> >
> > save("location");
> >
> > exit;
> >
> > };
> >
> >  # native SIP destinations are handled using our USRLOC DB
> >
> > if (!lookup("location"))
> >
> > {
> >
> > # Local Device Not Found Send To Gateway
> >
> > rewritehostport("<gateway>:5065");
> >
> > }
> >
> > };
> >
> > append_hf("P-hint: usrloc applied\r\n");
> >
> > route(1);
> >
> > }
> >
> > route[1]
> >
> > {
> >
> > # !! Nathelper
> >
> > # if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" &&
> > !search("^Route:"))
> >
> > # {
> >
> > # sl_send_reply("479", "We don't forward to private IP addresses");
> >
> > # exit;
> >
> > # };
> >
> > # NAT processing of replies; apply to all transactions (for example,
> >
> > # re-INVITEs from public to private UA are hard to identify as
> >
> > # NATed at the moment of request processing); look at replies
> >
> > t_on_reply("1");
> >
> > # send it out now; use stateful forwarding as it works reliably
> >
> > # even for UDP2TCP
> >
> > if (!t_relay()) {
> >
> > sl_reply_error();
> >
> > };
> >
> > }
> >
> > # !! Nathelper
> >
> > onreply_route[1]
> >
> > {
> >
> > if (nat_uac_test("31"))
> >
> > {
> >
> > # Allow RR-ed requests, as these may indicate that
> >
> > # a NAT-enabled proxy takes care of it; unless it is
> >
> > # a REGISTER
> >
> > xlog("Reply Behind a NAT");
> >
> > fix_nated_contact();
> >
> > force_rport(); # Add rport parameter to topmost Via
> >
> > #setbflag(6); # Mark as NATed
> >
> > };
> >
> > }
> >
> > _______________________________________________
> > Users mailing list
> > Users at lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> >
>
> --
> View this message in context:
> http://n2.nabble.com/MediaProxy-No-Audio-Problems-tp3911881p3913596.html
> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>
> _______________________________________________
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> Users at lists.opensips.org
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>
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