21 Sep
2006
21 Sep
'06
9:46 p.m.
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@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@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(a)voice-system.ro>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
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