Fwd: [Users] mediaproxy working, but not if asterisk is involved

Arne Van Theemsche arnevt at gmail.com
Mon Sep 25 09:35:31 CEST 2006


I had forgotten to cc the list

---------- Forwarded message ----------
From: Arne Van Theemsche <arnevt at gmail.com>
Date: 22-sep-2006 9:06
Subject: Re: [Users] mediaproxy working, but not if asterisk is involved
To: daniel at voice-system.ro

the problem is that I don't even see the "reply received"... So for some
reason the asterisk reply isn't passed through to the onreply_route. My
theory is that asterisk doesn't respect the reply parameters somewhere, but
it isn't clear to me where

arne


2006/9/22, Daniel-Constantin Mierla <daniel at voice-system.ro>:
>
> Do you get "using mediaproxy" message in the logs? If not, that the
> search() matches, I cannot sot right now what is wrong with the
> expression. But you can move t_on_reply("1") into if*method=="INVITE")
> statement and replace the search condition with if (status =~
> "(183)|(2[0-9][0-9])").
>
> See:
> http://voip-info.org/wiki/view/OpenSER+And+Mediaproxy
>
> Cheers,
> Daniel
>
>
> On 09/21/06 21:46, Arne Van Theemsche wrote:
> > below is the transaction of the failed mediaproxy invite. I allready
> > could tell that replies go through openser, but I don't see the reason
> > why ser doesn't see them as replies (and use the mediaproxy function).
> >
> > as you can see, the invite from <ip client> to <ip asterisk> (through
> > <ip OPENSER>, which is also ip of mediaproxy) goes in one direction
> > good (the ip in the SDP is changed from <ip client> to <ip openser>,
> > but the return path en the OK (with it's SDP) is not changed
> >
> > I did a tcpdump with a call between 2 clients, where the proxy works,
> > and the only difference I see is that in the reply of asterisk, there
> > is no rinstance field in the contact header
> >
> > thanks
> > arne
> >
> > U <ip client>:5060 -> <ip OPENSER>:5060
> >   INVITE sip:701@<sip domain>;transport=UDP SIP/2.0..From: "arne"
> > <sip:1002@<sip domain>>;tag=514a90c3-13c4-7a70a-1de331c0-5e4f..To:
> > "701"< sip:701 at sipgat
> >   e.evonet.be <http://e.evonet.be>>..Call-ID:
> > 1064dc44-514a90c3-13c4-7a70a-1de331be-529@<ip
> > <mailto: 1064dc44-514a90c3-13c4-7a70a-1de331be-529@%3Cip>
> > client>..CSeq: 1 INVITE..Via: SIP/2.0/UDP <ip
> > client>:5060;rport;branch=z9hG4bK-7a70a-1d
> >   e331c2-69dc..Max-Forwards: 70..Supported:
> > replaces,100rel,timer..Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL,
> > OPTIONS, INFO, PRACK..User-Agent: Swissvoice IP1
> >   0 SP v1.0.1 (Build 3) 3.0.5.1..Allow-Events: talk, hold,
> > conference..Contact: "arne" <sip:1002@<ip
> > client>:5060;transport=UDP>..Session-Expires: 1800..Content-
> >   Type: application/sdp..Content-Length: 246....v=0..o=rtp/1 501514
> > 501514 IN IP4 <ip client>..s=-..c=IN IP4 <ip client>..t=0 0..m=audio
> > 50000 RTP/AVP 18 0 8..
> >   a=fmtp:18 annexb=yes..a=ptime:40..a=SilenceSupp:on..a=rtpmap:18
> > g729/8000..a=rtpmap:0 pcmu/8000..a=rtpmap:8 pcma/8000..a=sendrecv..
> > #
> >
> > U <ip OPENSER>:5060 -> <ip asterisk>:5060
> >   INVITE sip:701@<sip domain>;transport=UDP SIP/2.0..Record-Route:
> > <sip:<ip OPENSER>;lr=on;ftag=514a90c3-13c4-7a70a-1de331c0-5e4f>..From:
> > "arne" < sip:1002 at si
> >   pgate.evonet.be
> > <http://pgate.evonet.be>>;tag=514a90c3-13c4-7a70a-1de331c0-5e4f..To:
> > "701"<sip:701@<sip domain>>..Call-ID:
> > 1064dc44-514a90c3-13c4-7a70a-1de331be-529@<ip
> > <mailto:1064dc44-514a90c3-13c4-7a70a-1de331be-529@%3Cip> client>..C
> >   Seq: 1 INVITE..Via: SIP/2.0/UDP <ip OPENSER>;branch=0..Via:
> > SIP/2.0/UDP <ip
> >
> client>:5060;rport=5060;branch=z9hG4bK-7a70a-1de331c2-69dc..Max-Forwards:
> > 69..Supp
> >   orted: replaces,100rel,timer..Allow: INVITE, ACK, BYE, REFER,
> > NOTIFY, CANCEL, OPTIONS, INFO, PRACK..User-Agent: Swissvoice IP10 SP
> > v1.0.1 (Build 3) 3.0.5.1..Allo
> >   w-Events: talk, hold, conference..Contact: "arne" <sip:1002@<ip
> > client>:5060;transport=UDP>..Session-Expires: 1800..Content-Type:
> > application/sdp..Content-Leng
> >   th: 246....v=0..o=rtp/1 501514 501514 IN IP4 <ip client>..s=-..c=IN
> > IP4 <ip OPENSER>..t=0 0..m=audio 60106 RTP/AVP 18 0 8..a=fmtp:18
> > annexb=yes..a=ptime:40..a
> >   =SilenceSupp:on..a=rtpmap:18 g729/8000..a=rtpmap:0
> > pcmu/8000..a=rtpmap:8 pcma/8000..a=sendrecv..
> > #
> >
> > U <ip asterisk>:5060 -> <ip OPENSER>:5060
> >   SIP/2.0 100 Trying..Via: SIP/2.0/UDP <ip
> > OPENSER>;branch=0;received=<ip OPENSER>..Via: SIP/2.0/UDP <ip
> > client>:5060;rport=5060;branch=z9hG4bK-7a70a-1de331c2-
> >   69dc..From: "arne" <sip:1002@<sip
> > domain>>;tag=514a90c3-13c4-7a70a-1de331c0-5e4f..To: "701"<sip:701@<sip
> > domain>>..Call-ID: 1064dc44-514a90c3-13c4-7a70
> >   a-1de331be-529@<ip <mailto:a-1de331be-529@%3Cip> client>..CSeq: 1
> > INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS,
> > BYE, REFER, SUBSCRIBE, NOTIFY..Contact: <sip:701@
> >   <ip asterisk>>..Content-Length: 0....
> > #
> >
> > U <ip OPENSER>:5060 -> <ip client>:5060
> >   SIP/2.0 100 Trying..Via: SIP/2.0/UDP <ip
> > client>:5060;rport=5060;branch=z9hG4bK-7a70a-1de331c2-69dc..From:
> > "arne" <sip:1002@<sip domain>>;tag=514a90c3-13c
> >   4-7a70a-1de331c0-5e4f..To: "701"<sip:701@<sip domain>>..Call-ID:
> > 1064dc44-514a90c3-13c4-7a70a-1de331be-529@<ip
> > <mailto:1064dc44-514a90c3-13c4-7a70a-1de331be-529@%3Cip>
> > client>..CSeq: 1 INVITE..User-Agent: Asteri
> >   sk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
> > NOTIFY..Contact: <sip:701@<ip asterisk>>..Content-Length: 0....
> > #
> >
> > U <ip asterisk>:5060 -> <ip OPENSER>:5060
> >   SIP/2.0 200 OK..Via: SIP/2.0/UDP <ip OPENSER>;branch=0;received=<ip
> > OPENSER>..Via: SIP/2.0/UDP <ip
> > client>:5060;rport=5060;branch=z9hG4bK-7a70a-1de331c2-69dc
> >   ..Record-Route: <sip:<ip
> > OPENSER>;lr=on;ftag=514a90c3-13c4-7a70a-1de331c0-5e4f>..From: "arne"
> > <sip:1002@<sip domain>>;tag=514a90c3-13c4-7a70a-1de331c0-5e4f
> >   ..To: "701"<sip:701@<sip domain>>;tag=as60ebd3fc..Call-ID:
> > 1064dc44-514a90c3-13c4-7a70a-1de331be-529@<ip
> > <mailto:1064dc44-514a90c3-13c4-7a70a-1de331be-529@%3Cip>
> > client>..CSeq: 1 INVITE..User-Agent: Asterisk PBX
> >   ..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
> > NOTIFY..Contact: <sip:701@<ip asterisk>>..Content-Type:
> > application/sdp..Content-Length: 188....v=
> >   0..o=root 26276 26276 IN IP4 <ip asterisk>..s=session..c=IN IP4 <ip
> > asterisk>..t=0 0..m=audio 13434 RTP/AVP 0 8..a=rtpmap:0
> > PCMU/8000..a=rtpmap:8 PCMA/8000..a=
> >   silenceSupp:off - - - -..
> > #
> >
> > U <ip OPENSER>:5060 -> <ip client>:5060
> >   SIP/2.0 200 OK..Via: SIP/2.0/UDP <ip
> >
> client>:5060;rport=5060;branch=z9hG4bK-7a70a-1de331c2-69dc..Record-Route:
> > <sip:<ip OPENSER>;lr=on;ftag=514a90c3-13c4-7a70
> >   a-1de331c0-5e4f>..From: "arne" <sip:1002@<sip
> > domain>>;tag=514a90c3-13c4-7a70a-1de331c0-5e4f..To: "701"<sip:701@<sip
> > domain>>;tag=as60ebd3fc..Call-ID:
> >   1064dc44-514a90c3-13c4-7a70a-1de331be-529@<ip
> > <mailto: 1064dc44-514a90c3-13c4-7a70a-1de331be-529@%3Cip>
> > client>..CSeq: 1 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK,
> > CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NO
> >   TIFY..Contact: <sip:701@<ip asterisk>>..Content-Type:
> > application/sdp..Content-Length: 188....v=0..o=root 26276 26276 IN IP4
> > <ip asterisk>..s=session..c=IN IP4
> >    <ip asterisk>..t=0 0..m=audio 13434 RTP/AVP 0 8..a=rtpmap:0
> > PCMU/8000..a=rtpmap:8 PCMA/8000..a=silenceSupp:off - - - -..
> > #
> >
> >
> >
> >
> >
> > 2006/9/21, Daniel-Constantin Mierla <daniel at voice-system.ro
> > <mailto:daniel at voice-system.ro>>:
> >
> >     Hello,
> >
> >     watch the network traffic with ngrep on your sip server. You can
> >     see the
> >     call flow which may help to identify the issue. You can paste it
> >     to the
> >     list and someone may give you hints.
> >
> >     Cheers,
> >     Daniel
> >
> >
> >     On 09/21/06 12:28, Arne Van Theemsche wrote:
> >     > hi
> >     >
> >     > my users subscribe with openser, en asterisk is used as
> connectivity
> >     > to pstn
> >     >
> >     > i am now installing a mediaproxy, for all users, so every call
> goes
> >     > via a mediaproxy.
> >     >
> >     > I'm doing this as follows (relevant statements only)
> >     >
> >     > in route
> >     >
> >     >         #I installed the t_on_reply here to be sure that every
> reply
> >     > gets parsed, but normally in the INVITE section should be enough?
> >     >         t_on_reply("1");
> >     >
> >     >         if (method==INVITE) {
> >     >                 use_media_proxy();
> >     >         }
> >     >
> >     >
> >     > onreply_route[1] {
> >     >         log(-3,"reply received");
> >     >         if (!search("^Content-Length:[ ]*0")) {
> >     >                 log(-3,"using mediaproxy");
> >     >                 use_media_proxy();
> >     >         };
> >     > }
> >     >
> >     >
> >     > the weird is, for all local users, this works fine, but as soon as
>
> >     > asterisk is involved, the reply doesn't get triggered (not
> >     seeing the
> >     > "reply received" either, only when disconnecting the call). The
> call
> >     > get's established fine, asterisk is sending media to the
> >     mediaproxy,
> >     > but  the SDP towards the calling phone is not modified (since the
> >     > onreply isn't triggered)
> >     >
> >     > am I missing something here?
> >     >
> >     > thanks
> >     > Arne
> >     >
> >     >
> >
> ------------------------------------------------------------------------
> >
> >     >
> >     > _______________________________________________
> >     > Users mailing list
> >     > Users at openser.org <mailto:Users at openser.org>
> >     > http://openser.org/cgi-bin/mailman/listinfo/users
> >     >
> >
> >
>
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