17 Dec
2004
17 Dec
'04
6:23 a.m.
the balancer is broken, the reply must follow back the path the request
had. Otherwise the proxies on the path should retransmit the request
until they get a reply or a timeout. Top most via header in reply must
include the ip address of the current proxy.
Daniel
On 12/16/04 22:27, Java Rockx wrote:
Hi all.
We have two ser-0.8.99-dev24 servers configured behind a load balancer, but we are having
record-route problems whereby an INVITE gets sent to an Asterisk server and the "100
Trying" reply
from Asterisk gets sent to the other SER proxy because the load balancer chooses to send
the 100
reply to the other ser proxy.
Below is an INVITE with a Record-Route and two Via headers. In this message the
68.84.225.30 IP is
the **virtual** public IP of the SER proxies.
Is the top VIA (ie, Via: SIP/2.0/UDP 68.84.226.30) the reason the "100 Trying"
message gets sent
to the other SIP proxy? We believe that if this VIA had the physical IP of the ser proxy
that sent
them message - then all would be fine.
We've played with record_route_preset("xxx.xxx.xxx.xxx") but that
doesn't seem to help with
regards to changing the top via.
Can anyone give some words of wisdom?
Regards,
Paul
U 10.2.20.21:5060 -> 10.3.0.23:5060
INVITE sip:4075551212@10.3.0.23:5060;user=phone SIP/2.0.
Record-Route: <sip:10.2.20.21;ftag=4d0c0449837b0fce;lr=on>.
Via: SIP/2.0/UDP 68.84.226.30;branch=z9hG4bK7277.557b17d3.0.
Via: SIP/2.0/UDP
192.168.0.195:5109;rport=5109;received=68.84.242.67;branch=z9hG4bKc5b09d33c79bfc1e.
From: <sip:4075551212@sip.mycompany.com;user=phone>;tag=4d0c0449837b0fce.
To: <sip:4075551212@sip.mycompany.com;user=phone>.
Contact: <sip:34075551212@68.84.242.67:5109;user=phone>.
Supported: replaces.
Proxy-Authorization: DIGEST username="4075551212",
realm="sip.mycompany.com", algorithm=MD5,
uri="sip:4075551212@sip.mycompany.com;user=phone",nonce="41c1f8a5c69796cff47ad459a19632f0d31105ee",
response="5613a719bd6a2b3b7315522611e09653".
Call-ID: 71cfc2d5a0fbdae1(a)192.168.0.195.
CSeq: 12663 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: 411.
X-host: 10.2.20.21 .
P-hint: 3 && Record-Route.
.
v=0.
o=3212514234 1 8000 IN IP4 192.168.0.195.
s=SIP Call.
c=IN IP4 68.84.226.26.
t=0 0.
m=audio 35338 RTP/AVP 98 18 15 4 2 9 0 8 101.
a=sendrecv.
a=rtpmap:98 iLBC/8000.
a=fmtp:98 mode=20.
a=rtpmap:18 G729/8000.
a=rtpmap:15 G728/8000.
a=rtpmap:4 G723/8000.
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.
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