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 <[email protected]> > > 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*[email protected]*<10002*[email protected]> > > * (b62884c7) --> **10001*[email protected]* <10001*[email protected]> > > *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*[email protected]*<10002*[email protected]> > > * (b62884c7) --> **10001*[email protected]* <10001*[email protected]> > > *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*[email protected]*<10002*[email protected]> > > * (b62884c7) --> **10001*[email protected]* <10001*[email protected]> > > *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*[email protected]'*<'10001*[email protected]'> > > *, 'from_uri': **'10002*[email protected]'* <'10002*[email protected]'>*, > > '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 > > [email protected] > > 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. > > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >
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