21 Jan
2005
21 Jan
'05
1:08 p.m.
Hi All.
Disregard my last cry for help.
I accidentally had a t_relay() for BYE messages before my loose_route() and that was
causing
obviously bad things to happen.
Regards,
Paul
--- Java Rockx <javarockx(a)yahoo.com> wrote:
Hi All.
I think I've identified a bug with how ser-0.9 handles BYE messages. It seems that
ser-0.9 is
not
honoring Record-Route headers when sending BYE messages.
Below is a complete SIP conversation between my GrandStream BT100 phone and a PSTN phone.
We use
a
3rd party for PSTN access and they utilize a Sonus box for SIP-to-TDM and a seperate SIP
proxy.
The actual problem is this; when a SIP phone hangs up **FIRST** the BYE message is sent
to their
Sonus Box directly rather than to their SIP proxy. Their Sonus box is listed as the
"Contact"
field in the SDP of the INVITE. The original INVITE was Record-Routed so shouldn't
the BYE be
sent
to this IP rather than to the other end-point directly?
When the phones go off-hook, our PSTN provider sends us a 200 OK which we ACK correctly,
but
then
the BYE is wrong.
Can anyone look at this conversation and tell me why the BYE is sent to 216.229.118.76
rather
than
216.229.127.60?
Regards,
Paul
IP LEGEND
---------------------------------------------------
68.84.100.10 - PUBLIC SIDE OF MY LAN FIREWALL
68.84.199.18 - IP ADDRESS OF MY SER SIP PROXY
216.229.127.60 - IP ADDRESS OF 3RD PARTY SIP PROXY
216.229.118.76 - IP ADDRESS OF 3RD PARTY SONUS BOX
U 2005/01/21 11:38:14.259621 68.84.100.10:63689 -> 68.84.199.18:5060
INVITE sip:3219392336@mycompany.com;user=phone SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.83:63689;branch=z9hG4bK67f005c3c771821e.
From: "Paul Hazlett"
<sip:3212514231@mycompany.com;user=phone>;tag=2440ace3b188639c.
To: <sip:3219392336@mycompany.com;user=phone>.
Contact: <sip:3212514231@192.168.0.83:63689;user=phone>.
Supported: replaces.
Call-ID: 870d0fdeaa2f0d0f(a)192.168.0.83.
CSeq: 27735 INVITE.
User-Agent: Grandstream BT100 1.0.5.16.
Max-Forwards: 70.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Content-Type: application/sdp.
Content-Length: 411.
.
v=0.
o=3212514231 0 8000 IN IP4 192.168.0.83.
s=SIP Call.
c=IN IP4 192.168.0.83.
t=0 0.
m=audio 39738 RTP/AVP 98 18 4 15 2 9 0 8 101.
a=sendrecv.
a=rtpmap:98 iLBC/8000.
a=fmtp:98 mode=20.
a=rtpmap:18 G729/8000.
a=rtpmap:4 G723/8000.
a=rtpmap:15 G728/8000.
a=rtpmap:2 G726-32/8000.
a=rtpmap:9 G722/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=ptime:40.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-11.
#
U 2005/01/21 11:38:14.264900 68.84.199.18:5060 -> 68.84.100.10:63689
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/UDP
192.168.0.83:63689;branch=z9hG4bK67f005c3c771821e;rport=63689;received=68.84.100.10.
From: "Paul Hazlett"
<sip:3212514231@mycompany.com;user=phone>;tag=2440ace3b188639c.
To:
<sip:3219392336@mycompany.com;user=phone>;tag=bf952ed189d8425c881b09485aa0b6f1.954a.
Call-ID: 870d0fdeaa2f0d0f(a)192.168.0.83.
CSeq: 27735 INVITE.
Proxy-Authenticate: Digest realm="mycompany.com",
nonce="41f13122784651d60bfa0cd3d5778587c1105f35".
Server: Sip eXpress Router (0.9.0 (i386/linux)).
Content-Length: 0.
.
#
U 2005/01/21 11:38:14.270933 68.84.100.10:63689 -> 68.84.199.18:5060
ACK sip:3219392336@mycompany.com;user=phone SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.83:63689;branch=z9hG4bK67f005c3c771821e.
From: "Paul Hazlett"
<sip:3212514231@mycompany.com;user=phone>;tag=2440ace3b188639c.
To:
<sip:3219392336@mycompany.com;user=phone>;tag=bf952ed189d8425c881b09485aa0b6f1.954a.
Contact: <sip:3212514231@192.168.0.83:63689;user=phone>.
Call-ID: 870d0fdeaa2f0d0f(a)192.168.0.83.
CSeq: 27735 ACK.
User-Agent: Grandstream BT100 1.0.5.16.
Max-Forwards: 70.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Content-Length: 0.
.
#
U 2005/01/21 11:38:14.288516 68.84.100.10:63689 -> 68.84.199.18:5060
INVITE sip:3219392336@mycompany.com;user=phone SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.83:63689;branch=z9hG4bK7779e54c00e334a6.
From: "Paul Hazlett"
<sip:3212514231@mycompany.com;user=phone>;tag=2440ace3b188639c.
To: <sip:3219392336@mycompany.com;user=phone>.
Contact: <sip:3212514231@192.168.0.83:63689;user=phone>.
Supported: replaces.
Proxy-Authorization: DIGEST username="3212514231",
realm="mycompany.com", algorithm=MD5,
uri="sip:3219392336@mycompany.com;user=phone",
nonce="41f13122784651d60bfa0cd3d5778587c1105f35",
response="2463041979c1bb36367b0d9d06bfa966".
Call-ID: 870d0fdeaa2f0d0f(a)192.168.0.83.
CSeq: 27736 INVITE.
User-Agent: Grandstream BT100 1.0.5.16.
Max-Forwards: 70.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Content-Type: application/sdp.
Content-Length: 411.
.
v=0.
o=3212514231 1 8000 IN IP4 192.168.0.83.
s=SIP Call.
c=IN IP4 192.168.0.83.
t=0 0.
m=audio 39738 RTP/AVP 98 18 4 15 2 9 0 8 101.
a=sendrecv.
a=rtpmap:98 iLBC/8000.
a=fmtp:98 mode=20.
a=rtpmap:18 G729/8000.
a=rtpmap:4 G723/8000.
a=rtpmap:15 G728/8000.
a=rtpmap:2 G726-32/8000.
a=rtpmap:9 G722/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=ptime:40.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-11.
#
U 2005/01/21 11:38:14.304423 68.84.199.18:5060 -> 68.84.100.10:63689
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP
192.168.0.83:63689;branch=z9hG4bK7779e54c00e334a6;rport=63689;received=68.84.100.10.
From: "Paul Hazlett"
<sip:3212514231@mycompany.com;user=phone>;tag=2440ace3b188639c.
To: <sip:3219392336@mycompany.com;user=phone>.
Call-ID: 870d0fdeaa2f0d0f(a)192.168.0.83.
CSeq: 27736 INVITE.
Server: Sip eXpress Router (0.9.0 (i386/linux)).
Content-Length: 0.
.
#
U 2005/01/21 11:38:14.304581 68.84.199.18:5060 -> 216.229.127.60:5060
INVITE sip:3219392336@216.229.127.60;user=phone SIP/2.0.
Record-Route: <sip:68.84.199.18;ftag=2440ace3b188639c;lr=on>.
Via: SIP/2.0/UDP 68.84.199.18;branch=z9hG4bKd55a.8a20c236.0.
Via: SIP/2.0/UDP
192.168.0.83:63689;rport=63689;received=68.84.100.10;branch=z9hG4bK7779e54c00e334a6.
From: "Paul Hazlett"
<sip:3212514231@mycompany.com;user=phone>;tag=2440ace3b188639c.
To: <sip:3219392336@mycompany.com;user=phone>.
Contact: <sip:3212514231@68.84.100.10:63689;user=phone>.
Supported: replaces.
Proxy-Authorization: DIGEST username="3212514231",
realm="mycompany.com", algorithm=MD5,
uri="sip:3219392336@mycompany.com;user=phone",
nonce="41f13122784651d60bfa0cd3d5778587c1105f35",
response="2463041979c1bb36367b0d9d06bfa966".
Call-ID: 870d0fdeaa2f0d0f(a)192.168.0.83.
CSeq: 27736 INVITE.
User-Agent: Grandstream BT100 1.0.5.16.
Max-Forwards: 16.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Content-Type: application/sdp.
Content-Length: 412.
.
v=0.
o=3212514231 1 8000 IN IP4 192.168.0.83.
s=SIP Call.
c=IN IP4 68.84.199.18.
t=0 0.
m=audio 36096 RTP/AVP 98 18 4 15 2 9 0 8 101.
a=sendrecv.
a=rtpmap:98 iLBC/8000.
a=fmtp:98 mode=20.
a=rtpmap:18 G729/8000.
a=rtpmap:4 G723/8000.
a=rtpmap:15 G728/8000.
a=rtpmap:2 G726-32/8000.
a=rtpmap:9 G722/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=ptime:40.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-11.
#
U 2005/01/21 11:38:14.370297 216.229.127.60:5060 -> 68.84.199.18:5060
SIP/2.0 100 Try.
Via: SIP/2.0/UDP 68.84.199.18;branch=z9hG4bKd55a.8a20c236.0.
Via: SIP/2.0/UDP
192.168.0.83:63689;rport=63689;received=68.84.100.10;branch=z9hG4bK7779e54c00e334a6.
To: <sip:3219392336@mycompany.com;user=phone>.
=== message truncated ===
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