Thread: [OpenSIPStack] opensbc media proxy?
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joegenbaclor
From: Jeremy A <je...@el...> - 2007-11-18 19:42:04
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Hello, In doing tests on 1.1.4 opensbc I notice that the media stream is not being proxied. In my test setup I have a remote SIP registrar, and a local Linksys phone. Normally the phone registers with the remote proxy. I changed the phone setup to use an opensbc instance by setting the 'Outbound Proxy:' value on the phone. The following fields on the phone are set to 'no' or are blank. NAT Mapping Enable: SIP Proxy-Require: Handle VIA received: Handle VIA rport: Insert VIA received: Insert VIA rport: Substitute VIA Addr: Send Resp To Src Port: The phone and the opensbc instance are on routable addresses The phone uses the opensbc in its SIP dialog as expected. My assumption was that the opensbc instance would automatically proxy the RTP audio as well as manage the SIP conversation. Is this assumption correct? Is there any special setting I need to make to the phone or openSBC to allow audio proxying? Thanks Jeremy |
From: Joegen E. B. <joe...@gm...> - 2007-11-18 23:55:11
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OpenSBC will stay away from proxying media if the endpoints in the call are both using public addresses. You can tell OpenSBC to always proxy media in all cases by setting "Always Proxy Media" in HTTP admin "General Parameters" section. hth. Joegen Jeremy A wrote: > Hello, > > In doing tests on 1.1.4 opensbc I notice that the media stream is not > being proxied. > > In my test setup I have a remote SIP registrar, and a local Linksys > phone. Normally the phone registers with the remote proxy. I changed the > phone setup to use an opensbc instance by setting the 'Outbound Proxy:' > value on the phone. > > The following fields on the phone are set to 'no' or are blank. > > NAT Mapping Enable: > SIP Proxy-Require: > > Handle VIA received: > Handle VIA rport: > Insert VIA received: > Insert VIA rport: > Substitute VIA Addr: > Send Resp To Src Port: > > The phone and the opensbc instance are on routable addresses > > The phone uses the opensbc in its SIP dialog as expected. My assumption > was that the opensbc instance would automatically proxy the RTP audio as > well as manage the SIP conversation. > > Is this assumption correct? > > Is there any special setting I need to make to the phone or openSBC to > allow audio proxying? > > Thanks > > Jeremy > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Microsoft > Defy all challenges. Microsoft(R) Visual Studio 2005. > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > |
From: Jeremy A <je...@el...> - 2007-11-19 01:04:46
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Joegen E. Baclor wrote: > OpenSBC will stay away from proxying media if the endpoints in the call > are both using public addresses. You can tell OpenSBC to always proxy > media in all cases by setting "Always Proxy Media" in HTTP admin > "General Parameters" section. > > I have that flag ticked. There are two distinct cases. 1. The phone makes an outgoing call (actually to voicemail on the remote SIP server). All RTP traffic goes direct from the remote host to the local phone bypassing opensbc. it is bidirectional. 2. I make an incoming call from the remote SIP server. The call is handled by opensbc and the local phone rings. When the phone is answered the SDP handshake managed by opensbc directs both the remote SIP server and the local phone to send packets to 0.0.31.64 - so no sound on the phone and no traffic through opensbc. It would appear that the note "Enable proxying media on all incoming call" on the "General Parameters" "Always Proxy Media" is literal - only incoming calls are proxied. It would also appear that there is a bug in the code setting up the RTP proxy. Possibly this is due to me running on fully routable addresses? Jeremy |
From: <jo...@op...> - 2007-11-19 01:18:13
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Can you send a level 5 log for this case? OpenSBC giving 0.0.31.64 as the media address is strange. Jeremy A wrote: > Joegen E. Baclor wrote: > >> OpenSBC will stay away from proxying media if the endpoints in the call >> are both using public addresses. You can tell OpenSBC to always proxy >> media in all cases by setting "Always Proxy Media" in HTTP admin >> "General Parameters" section. >> >> >> > I have that flag ticked. There are two distinct cases. > > 1. The phone makes an outgoing call (actually to voicemail on the remote > SIP server). All RTP traffic goes direct from the remote host to the > local phone bypassing opensbc. it is bidirectional. > > 2. I make an incoming call from the remote SIP server. The call is > handled by opensbc and the local phone rings. When the phone is answered > the SDP handshake managed by opensbc directs both the remote SIP server > and the local phone to send packets to 0.0.31.64 - so no sound on the > phone and no traffic through opensbc. > > It would appear that the note "Enable proxying media on all incoming > call" on the "General Parameters" "Always Proxy Media" is literal - only > incoming calls are proxied. > > It would also appear that there is a bug in the code setting up the RTP > proxy. Possibly this is due to me running on fully routable addresses? > > Jeremy > > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Microsoft > Defy all challenges. Microsoft(R) Visual Studio 2005. > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > > |