[Serusers] HELP: Via in multi-host SER configuration

[Java Rockx]

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 at 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 at sip.mycompany.com;user=phone>;tag=4d0c0449837b0fce.
To: <sip:4075551212 at sip.mycompany.com;user=phone>.
Contact: <sip:34075551212 at 68.84.242.67:5109;user=phone>.
Supported: replaces.
Proxy-Authorization: DIGEST username="4075551212", realm="sip.mycompany.com", algorithm=MD5,
uri="sip:4075551212 at sip.mycompany.com;user=phone",nonce="41c1f8a5c69796cff47ad459a19632f0d31105ee",
response="5613a719bd6a2b3b7315522611e09653".
Call-ID: 71cfc2d5a0fbdae1 at 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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