[OpenSIPS-Users] t_relay() not relaying payload
Thamer Alharbash
talharbash at gmail.com
Wed Feb 3 23:19:43 CET 2010
Hi Bogdan,
Sorry for not getting back sooner. I've updated my config a bit. I'm
including what our reinvite handling looks like and the two reinvites
that pass through opensips. The second one as you can see has no
payload (ngrep shows ...) I have verified this as well under wireshark.
if (has_totag()) {
# sequential request within a dialog should
# take the path determined by record-routing
if (loose_route()) {
if (is_method("BYE")) {
setflag(1); # do accounting ...
setflag(3); # ... even if the
transaction fails
} else if (is_method("INVITE")) {
# even if in most of the cases is
useless, do RR for
# re-INVITEs also, as some buggy
clients do change route set
# during the dialog.
record_route_preset
("proxy.ip.address.here");
}
# route it out to whatever destination was
set by loose_route()
# in $du (destination URI).
route(1);
...
route[1] {
# for INVITEs enable some additional helper routes
if (is_method("INVITE")) {
t_on_branch("1");
t_on_reply("1");
t_on_failure("1");
if(has_body("application/sdp")) {
rtpproxy_offer("frc","proxy.ip.address.here");
xlog ("Setting rtpproxy_offer");
}
if (isbflagset(6)) {
fix_nated_contact();
}
}
-- FIRST REINVITE
U carrier.ip.address.here:5060 -> our.proxy.ip.address:5060
INVITE sip:1003 at uac.ip.address.here:5060;transport=udp;user=phone SIP/
2.0.
Via: SIP/2.0/UDP carrier.ip.address.here:
5060;branch=z9hG4bK2kspjh305gqgrfskk5s0sbk9m0g10.1.
Call-ID: 3110d9027c77f4e246f17ef1f5b735e0 at 192.168.2.12.
From: <sip:
2165928121 at our.proxy.hostname.here;user=phone>;tag=SD1ko0299-324092c7
+1+ad440023+8233f214.
To: "Thamer Al-Harbash" <sip:
1003 at our.proxy.hostname.here;user=phone>;tag=83134be1983195b6.
CSeq: 878247939 INVITE.
Expires: 180.
Min-SE: 1800.
Session-Expires: 1800;refresher=uac.
Supported: 100rel,timer.
Max-Forwards: 69.
Contact: <sip:2165928121 at our.proxy.ip.address:5060;transport=udp>.
Content-Type: application/sdp.
Content-Length: 216.
Route: <sip:carrier.ip.address.here;lr=on>.
.
v=0.
o=- 3474221042 3474221054 IN IP4 carrier.ip.address.here.
s=-.
c=IN IP4 carrier.ip.address.here.
t=0 0.
m=audio 10044 RTP/AVP 0 101.
a=sendrecv.
a=ptime:20.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
U our.proxy.ip.address:5060 -> carrier.ip.address.here:5060
SIP/2.0 100 Giving a try.
Via: SIP/2.0/UDP carrier.ip.address.here:
5060;branch=z9hG4bK2kspjh305gqgrfskk5s0sbk9m0g10.1.
Call-ID: 3110d9027c77f4e246f17ef1f5b735e0 at 192.168.2.12.
From: <sip:
2165928121 at our.proxy.hostname.here;user=phone>;tag=SD1ko0299-324092c7
+1+ad440023+8233f214.
To: "Thamer Al-Harbash" <sip:
1003 at our.proxy.hostname.here;user=phone>;tag=83134be1983195b6.
CSeq: 878247939 INVITE.
Server: OpenSIPS (1.6.0-notls (i386/linux)).
Content-Length: 0.
.
U our.proxy.ip.address:5060 -> uac.ip.address.here:5060
INVITE sip:1003 at uac.ip.address.here:5060;transport=udp;user=phone SIP/
2.0.
Record-Route: <sip:our.proxy.ip.address;lr=on>.
Via: SIP/2.0/UDP our.proxy.ip.address;branch=z9hG4bK20d7.90192965.0.
Via: SIP/2.0/UDP carrier.ip.address.here:
5060;branch=z9hG4bK2kspjh305gqgrfskk5s0sbk9m0g10.1.
Call-ID: 3110d9027c77f4e246f17ef1f5b735e0 at 192.168.2.12.
From: <sip:
2165928121 at our.proxy.hostname.here;user=phone>;tag=SD1ko0299-324092c7
+1+ad440023+8233f214.
To: "Thamer Al-Harbash" <sip:
1003 at our.proxy.hostname.here;user=phone>;tag=83134be1983195b6.
CSeq: 878247939 INVITE.
Expires: 180.
Min-SE: 1800.
Session-Expires: 1800;refresher=uac.
Supported: 100rel,timer.
Max-Forwards: 68.
Contact: <sip:2165928121 at carrier.ip.address.here:5060;transport=udp>.
Content-Type: application/sdp.
Content-Length: 217.
.
v=0.
o=- 3474221042 3474221054 IN IP4 carrier.ip.address.here.
s=-.
c=IN IP4 our.proxy.ip.address.
t=0 0.
m=audio 32104 RTP/AVP 0 101.
a=sendrecv.
a=ptime:20.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
-- SECOND REINVITE
U carrier.ip.address.here:5060 -> our.proxy.ip.address:5060
INVITE sip:1003 at 192.168.2.12:5060;transport=udp;user=phone SIP/2.0.
Via: SIP/2.0/UDP carrier.ip.address.here:
5060;branch=z9hG4bK2kspjh305gqgrfskk5s0sbk9m0020.1.
Call-ID: 3110d9027c77f4e246f17ef1f5b735e0 at 192.168.2.12.
From: <sip:
2165928121 at our.proxy.hostname.here;user=phone>;tag=SD1ko0299-324092c7
+1+ad440023+8233f214.
To: "Thamer Al-Harbash" <sip:
1003 at our.proxy.hostname.here;user=phone>;tag=83134be1983195b6.
CSeq: 878247940 INVITE.
Expires: 180.
Min-SE: 1800.
Session-Expires: 1800;refresher=uac.
Supported: 100rel,timer.
Max-Forwards: 69.
Contact: <sip:2165928121 at carrier.ip.address.here:5060;transport=udp>.
Content-Type: application/sdp.
Content-Length: 216.
Route: <sip:our.proxy.ip.address;lr=on>.
.
v=0.
o=- 3474221042 3474221054 IN IP4 carrier.ip.address.here.
s=-.
c=IN IP4 carrier.ip.address.here.
t=0 0.
m=audio 10044 RTP/AVP 0 101.
a=sendrecv.
a=ptime:20.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
U our.proxy.ip.address:5060 -> carrier.ip.address.here:5060
SIP/2.0 100 Giving a try.
Via: SIP/2.0/UDP carrier.ip.address.here:
5060;branch=z9hG4bK2kspjh305gqgrfskk5s0sbk9m0020.1.
Call-ID: 3110d9027c77f4e246f17ef1f5b735e0 at 192.168.2.12.
From: <sip:
2165928121 at our.proxy.hostname.here;user=phone>;tag=SD1ko0299-324092c7
+1+ad440023+8233f214.
To: "Thamer Al-Harbash" <sip:
1003 at our.proxy.hostname.here;user=phone>;tag=83134be1983195b6.
CSeq: 878247940 INVITE.
Server: OpenSIPS (1.6.0-notls (i386/linux)).
Content-Length: 0.
.
U our.proxy.ip.address:5060 -> uac.ip.address.here:5060
....
U our.proxy.ip.address:5060 -> carrier.ip.address.here:5060
SIP/2.0 408 Request Timeout.
Via: SIP/2.0/UDP carrier.ip.address.here:
5060;branch=z9hG4bK2kspjh305gqgrfskk5s0sbk9m0020.1.
Call-ID: 3110d9027c77f4e246f17ef1f5b735e0 at 192.168.2.12.
From: <sip:
2165928121 at our.proxy.hostname.here;user=phone>;tag=SD1ko0299-324092c7
+1+ad440023+8233f214.
To: "Thamer Al-Harbash" <sip:
1003 at our.proxy.hostname.here;user=phone>;tag=83134be1983195b6.
CSeq: 878247940 INVITE.
Server: OpenSIPS (1.6.0-notls (i386/linux)).
Content-Length: 0.
.
On 1-Feb-10, at 8:55 AM, Bogdan-Andrei Iancu wrote:
> Hi Thamer,
>
> Could you post the first re-INVITE (received and sent) to see what
> could
> be the problem?
>
> Regards,
> Bogdan
>
> Thamer Alharbash wrote:
>> We currently have opensips setup to route through another carrier for
>> certain calls. All signaling and media works well except for
>> reinvites.
>>
>> if (has_totag()) {
>>
>> # sequential request within a dialog should
>> # take the path determined by record-routing
>>
>> if (loose_route()) {
>> if (is_method("BYE")) {
>> setflag(1); # do accounting ...
>> setflag(3); # ... even if the
>> transaction fails
>> } else if (is_method("INVITE")) {
>> # even if in most of the cases is
>> useless, do RR for
>> # re-INVITEs alos, as some buggy
>> clients do change route set
>> # during the dialog.
>> record_route_preset("<hidden>");
>> }
>> fix_nated_contact();
>> t_relay();
>> ...
>>
>> The first reinvite passes through fine with the nated contact fixed
>> for the contact field. The second reinvite does not get relayed
>> correctly. Instead a udp packet with no SIP payload at all is sent to
>> the UA. We can't find any particular error in the debug log.
>>
>> Does anyone have thoughts on this?
>>
>>
>
>
> --
> Bogdan-Andrei Iancu
> www.voice-system.ro
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Thamer
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