[Users] mediaproxy working, but not if asterisk is involved
Arne Van Theemsche
arnevt at gmail.com
Thu Sep 21 20:46:08 CEST 2006
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>..Call-ID:
1064dc44-514a90c3-13c4-7a70a-1de331be-529@<ipclient>..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>;tag=514a90c3-13c4-7a70a-1de331c0-5e4f..To:
"701"<sip:701@<sip domain>>..Call-ID:
1064dc44-514a90c3-13c4-7a70a-1de331be-529@<ip 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 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 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 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 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>:
>
> 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
> > http://openser.org/cgi-bin/mailman/listinfo/users
> >
>
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