[OpenSIPS-Devel] Devel Digest, Vol 71, Issue 42

kaushik parmar androidjpc0 at gmail.com
Tue May 20 17:08:53 CEST 2014


Thank you Razvan,


As you said rtpproxy2 will send rtp to rtpproxy1. How can i implement this?

What are configuration in opensips or rtpproxy to make this possible?




On Tue, May 20, 2014 at 3:30 PM, <devel-request at lists.opensips.org> wrote:

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> Today's Topics:
>
>    1. [opensips] UAC_REGISTRANT module: fixed expiration time
>       calculation (#236) (Nick Altmann)
>    2. Re: [opensips] uac_registrant and external "expires"
>       parameter (in 200OK) (#217) (Nick Altmann)
>    3. Re: [opensips] UAC_REGISTRANT module: fixed expiration time
>       calculation (#234) (Nick Altmann)
>    4. [opensips] RTPPROXY module: compatibility with    topology
>       hiding (#237) (Nick Altmann)
>    5. Re: [opensips] RTPPROXY module: compatibility with topology
>       hiding (#237) (Walter Doekes)
>    6. Re: [opensips] RTPPROXY module: compatibility with topology
>       hiding (#237) (Walter Doekes)
>    7. Send rtp packet between two rtpproxy server (kaushik parmar)
>    8. Re: Send rtp packet between two rtpproxy server (R?zvan Crainea)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Mon, 19 May 2014 23:01:24 -0700
> From: Nick Altmann <notifications at github.com>
> Subject: [OpenSIPS-Devel] [opensips] UAC_REGISTRANT module: fixed
>         expiration time calculation (#236)
> To: devel at lists.opensips.org
> Message-ID: <OpenSIPS/opensips/pull/236 at github.com>
> Content-Type: text/plain; charset="utf-8"
>
>
> You can merge this Pull Request by running:
>
>   git pull https://github.com/nikbyte/nikbyte-opensips uac_registrant
>
> Or you can view, comment on it, or merge it online at:
>
>   https://github.com/OpenSIPS/opensips/pull/236
>
> -- Commit Summary --
>
>   * UAC_REGISTRANT module: fixed expiration time calculation
>
> -- File Changes --
>
>     M modules/uac_registrant/registrant.c (13)
>
> -- Patch Links --
>
> https://github.com/OpenSIPS/opensips/pull/236.patch
> https://github.com/OpenSIPS/opensips/pull/236.diff
>
> ---
> Reply to this email directly or view it on GitHub:
> https://github.com/OpenSIPS/opensips/pull/236
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> ------------------------------
>
> Message: 2
> Date: Mon, 19 May 2014 23:02:59 -0700
> From: Nick Altmann <notifications at github.com>
> Subject: Re: [OpenSIPS-Devel] [opensips] uac_registrant and external
>         "expires" parameter (in 200OK) (#217)
> To: devel at lists.opensips.org
> Message-ID: <OpenSIPS/opensips/issues/217/43589176 at github.com>
> Content-Type: text/plain; charset="utf-8"
>
> Recreated pull request. The proper one:
> https://github.com/OpenSIPS/opensips/pull/236
>
> ---
> Reply to this email directly or view it on GitHub:
> https://github.com/OpenSIPS/opensips/issues/217#issuecomment-43589176
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> ------------------------------
>
> Message: 3
> Date: Mon, 19 May 2014 23:03:05 -0700
> From: Nick Altmann <notifications at github.com>
> Subject: Re: [OpenSIPS-Devel] [opensips] UAC_REGISTRANT module: fixed
>         expiration time calculation (#234)
> To: devel at lists.opensips.org
> Message-ID:
>         <OpenSIPS/opensips/pull/234/issue_event/122760818 at github.com>
> Content-Type: text/plain; charset="utf-8"
>
> Closed #234.
>
> ---
> Reply to this email directly or view it on GitHub:
> https://github.com/OpenSIPS/opensips/pull/234#event-122760818
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>
> ------------------------------
>
> Message: 4
> Date: Mon, 19 May 2014 23:06:53 -0700
> From: Nick Altmann <notifications at github.com>
> Subject: [OpenSIPS-Devel] [opensips] RTPPROXY module: compatibility
>         with    topology hiding (#237)
> To: devel at lists.opensips.org
> Message-ID: <OpenSIPS/opensips/pull/237 at github.com>
> Content-Type: text/plain; charset="utf-8"
>
> RTPPROXY module: workaround to run rtpproxy when we use dialog topology
> hiding and thus we have not 2nd via in reply
>
> It fixes autobridging for rtpproxy when using topology hiding.
> The problem is reply doesn&#39;t have second via.
>
> We used this patch on production system about 7 months without any
> problems.
> You can merge this Pull Request by running:
>
>   git pull https://github.com/nikbyte/nikbyte-opensips rtpproxy
>
> Or you can view, comment on it, or merge it online at:
>
>   https://github.com/OpenSIPS/opensips/pull/237
>
> -- Commit Summary --
>
>   * RTPPROXY module: workaround to run rtpproxy when we use dialog
> topology hiding and thus we have not 2nd via in reply
>
> -- File Changes --
>
>     M modules/rtpproxy/rtpproxy.c (22)
>
> -- Patch Links --
>
> https://github.com/OpenSIPS/opensips/pull/237.patch
> https://github.com/OpenSIPS/opensips/pull/237.diff
>
> ---
> Reply to this email directly or view it on GitHub:
> https://github.com/OpenSIPS/opensips/pull/237
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> ------------------------------
>
> Message: 5
> Date: Tue, 20 May 2014 00:00:54 -0700
> From: Walter Doekes <notifications at github.com>
> Subject: Re: [OpenSIPS-Devel] [opensips] RTPPROXY module:
>         compatibility with topology hiding (#237)
> To: devel at lists.opensips.org
> Message-ID: <OpenSIPS/opensips/pull/237/r12830089 at github.com>
> Content-Type: text/plain; charset="utf-8"
>
> > @@ -3229,6 +3229,8 @@ force_rtp_proxy(struct sip_msg* msg, char* str1,
> char* str2, char *setid, char *
> >       struct force_rtpp_args args;
> >       struct force_rtpp_args *ap;
> >       union sockaddr_union to;
> > +     struct socket_info *si;
> > +     struct dlg_cell * dlg;
>
> Please stay consistent with local code style: join the `*` to the `dlg`.
>
> ---
> Reply to this email directly or view it on GitHub:
> https://github.com/OpenSIPS/opensips/pull/237/files#r12830089
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> ------------------------------
>
> Message: 6
> Date: Tue, 20 May 2014 00:01:40 -0700
> From: Walter Doekes <notifications at github.com>
> Subject: Re: [OpenSIPS-Devel] [opensips] RTPPROXY module:
>         compatibility with topology hiding (#237)
> To: devel at lists.opensips.org
> Message-ID: <OpenSIPS/opensips/pull/237/r12830105 at github.com>
> Content-Type: text/plain; charset="utf-8"
>
> >                                       } else {
> > -                                             LM_ERR("can't extract 2nd
> via found reply\n");
> > +                                             if (parse_headers(msg,
> HDR_VIA2_F, 0) != -1 &&
>
> You could make the `} else {` above an `} else if {`, that saves you some
> indentation.
>
> Feel free to squash and push -f.
>
> ---
> Reply to this email directly or view it on GitHub:
> https://github.com/OpenSIPS/opensips/pull/237/files#r12830105
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> ------------------------------
>
> Message: 7
> Date: Tue, 20 May 2014 12:47:27 +0530
> From: kaushik parmar <androidjpc0 at gmail.com>
> Subject: [OpenSIPS-Devel] Send rtp packet between two rtpproxy server
> To: devel at lists.opensips.org
> Message-ID:
>         <CAGzPE1-QJdJ4-LKhwn4rVWeEFGUJQ0RZG92kquWyw+c=
> vhpeqQ at mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> Hello,
>
> Is it possible to send media rtp packets from one rtpproxy server to
> another rtpproxy server?
>
> In my scenario , i am registering voip account via opensips proxy server.
> We have rtpproxy and opensips server hosted on same place. opensips changes
> c= and m= lines of SDP accordingly but when packet goes to voip switch ,
> its rtpproxy server also changes SDP. So Peer1 is sending packets to hout
> hosted rtpproxy server and peer2 sending rtp packets to voip switch's
> rtpproxy server.
>
>
> SIP packets : Peer <--> opensips <--> asterisk
>
> RTP packets peer1 <--> rtpproxy1      =X=      rtpproxy2 <-->peer2
>
> here no connection between rtpproxy1 and rtpproxy 2 so no media transfer
> between peer1 and peer2. Please help to solve this problem.
>
> --
> Kind regards,
>
> Kaushik Parmar
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> ------------------------------
>
> Message: 8
> Date: Tue, 20 May 2014 10:34:42 +0300
> From: R?zvan Crainea <razvan at opensips.org>
> Subject: Re: [OpenSIPS-Devel] Send rtp packet between two rtpproxy
>         server
> To: devel at lists.opensips.org
> Message-ID: <537B0592.8060901 at opensips.org>
> Content-Type: text/plain; charset=UTF-8; format=flowed
>
> Hi, Kaushik!
>
> I don't see any problem with your scenario, it should work without any
> issues - peer2 should send RTP to rtpproxy2 which relays them to
> rtpproy1 and then peer1, and the other way around.
> Are you sure both proxies advertise the correct IP in the SDP, for both
> INVITE and 200 OK?
>
> Best regards,
>
> PS: this is not a devel question. Please register on
> users at lists.opensips.org and move the thread there.
>
> Razvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
>
> On 05/20/2014 10:17 AM, kaushik parmar wrote:
> > Hello,
> >
> > Is it possible to send media rtp packets from one rtpproxy server to
> > another rtpproxy server?
> >
> > In my scenario , i am registering voip account via opensips proxy
> > server. We have rtpproxy and opensips server hosted on same place.
> > opensips changes c= and m= lines of SDP accordingly but when packet goes
> > to voip switch , its rtpproxy server also changes SDP. So Peer1 is
> > sending packets to hout hosted rtpproxy server and peer2 sending rtp
> > packets to voip switch's rtpproxy server.
> >
> >
> > SIP packets : Peer <--> opensips <--> asterisk
> >
> > RTP packets peer1 <--> rtpproxy1      =X=      rtpproxy2 <-->peer2
> > here no connection between rtpproxy1 and rtpproxy 2 so no media transfer
> > between peer1 and peer2. Please help to solve this problem.
> >
> > --
> > Kind regards,
> >
> > Kaushik Parmar
> >
> >
> > _______________________________________________
> > Devel mailing list
> > Devel at lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
> >
>
>
>
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> End of Devel Digest, Vol 71, Issue 42
> *************************************
>



-- 
Kind regards,

Kaushik Parmar
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