[OpenSIPS-Users] BYE in a load balancing with asterisk scenario

Fri Oct 8 13:14:56 CEST 2010

Hi!

I also have another problem: since opensips is configured as asterisk 
peer is *'s config, the BYE asterisk sends does not get to the load 
balanced UA when it is NATed. Obviously, this is because there is no Via 
header in the BYE message with received and rport fields (sip trace 
below). Is there any way to solve this? Somehow remember the Via header 
for this call and add it to BYE when it passes through opensips (and 
matches the call)?
Here is the trace from *'s point of view:

<--- SIP read from xxx.yyy.zzz.90:5060 --->
INVITE sip:to at xxx.yyy.zzz.90 SIP/2.0
Record-Route: <sip:xxx.yyy.zzz.90;lr;ftag=dwage;did=643.3986f963>
Via: SIP/2.0/UDP xxx.yyy.zzz.90;branch=z9hG4bK931f.33d7e6f4.0
Via: SIP/2.0/UDP 
192.168.128.205;received=cli.ent.i.p;rport=5060;branch=z9hG4bKhkaqxvra
Max-Forwards: 69
Proxy-Authorization: Digest 
username="from",realm="asterisk",nonce="51e71e69",uri="sip:to at xxx.yyy.zzz.90",response="c643612
2f92fe7cf08006dc154705638",algorithm=MD5
To: <sip:to at xxx.yyy.zzz.90>
From: "aaa" <sip:from at xxx.yyy.zzz.90>;tag=dwage
Call-ID: vpdjrkrfbsmjkvy at aalexandrov.site
CSeq: 131 INVITE
Contact: <sip:from at 192.168.128.205>
Content-Type: application/sdp
Allow: 
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
Supported: replaces,norefersub,100rel
User-Agent: Twinkle/1.4.2
Content-Length: 313

v=0
o=twinkle 830887642 558243211 IN IP4 192.168.128.205
s=-
c=IN IP4 192.168.128.205
t=0 0
m=audio 8000 RTP/AVP 98 97 8 0 3 101
a=rtpmap:98 speex/16000
a=rtpmap:97 speex/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

<--- Transmitting (NAT) to xxx.yyy.zzz.90:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 
xxx.yyy.zzz.90;branch=z9hG4bK931f.33d7e6f4.0;received=xxx.yyy.zzz.90
Via: SIP/2.0/UDP 
192.168.128.205;received=cli.ent.i.p;rport=5060;branch=z9hG4bKhkaqxvra
Record-Route: <sip:xxx.yyy.zzz.90;lr;ftag=dwage;did=643.3986f963>
From: "aaa" <sip:from at xxx.yyy.zzz.90>;tag=dwage
To: <sip:to at xxx.yyy.zzz.90>
Call-ID: vpdjrkrfbsmjkvy at aalexandrov.site
CSeq: 131 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:to at xxx.yyy.zzz.94>
Content-Length: 0

<--- Reliably Transmitting (NAT) to xxx.yyy.zzz.90:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
xxx.yyy.zzz.90;branch=z9hG4bK931f.33d7e6f4.0;received=xxx.yyy.zzz.90
Via: SIP/2.0/UDP 
192.168.128.205;received=cli.ent.i.p;rport=5060;branch=z9hG4bKhkaqxvra
Record-Route: <sip:xxx.yyy.zzz.90;lr;ftag=dwage;did=643.3986f963>
From: "aaa" <sip:from at xxx.yyy.zzz.90>;tag=dwage
To: <sip:to at xxx.yyy.zzz.90>;tag=as671f3653
Call-ID: vpdjrkrfbsmjkvy at aalexandrov.site
CSeq: 131 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:to at xxx.yyy.zzz.94>
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 29029 29029 IN IP4 xxx.yyy.zzz.94
s=session
c=IN IP4 xxx.yyy.zzz.94
t=0 0
m=audio 15576 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<--- SIP read from xxx.yyy.zzz.90:5060 --->
ACK sip:to at xxx.yyy.zzz.94 SIP/2.0
Via: SIP/2.0/UDP xxx.yyy.zzz.90;branch=z9hG4bK931f.33d7e6f4.2
Via: SIP/2.0/UDP 
192.168.128.205;received=cli.ent.i.p;rport=5060;branch=z9hG4bKgvysatjo
Max-Forwards: 69
Proxy-Authorization: Digest 
username="from",realm="asterisk",nonce="51e71e69",uri="sip:to at xxx.yyy.zzz.90",response="c6436122f92fe7cf08006dc154705638",algorithm=MD5
To: <sip:to at xxx.yyy.zzz.90>;tag=as671f3653
From: "aaa" <sip:from at xxx.yyy.zzz.90>;tag=dwage
Call-ID: vpdjrkrfbsmjkvy at aalexandrov.site
CSeq: 131 ACK
User-Agent: Twinkle/1.4.2
Content-Length: 0

Reliably Transmitting (NAT) to xxx.yyy.zzz.90:5060:
BYE sip:from at 192.168.128.205 SIP/2.0
Via: SIP/2.0/UDP xxx.yyy.zzz.94:5060;branch=z9hG4bK1890cc40;rport
Route: <sip:xxx.yyy.zzz.90;lr;ftag=dwage;did=643.3986f963>
From: <sip:to at xxx.yyy.zzz.90>;tag=as671f3653
To: "aaa" <sip:from at xxx.yyy.zzz.90>;tag=dwage
Call-ID: vpdjrkrfbsmjkvy at aalexandrov.site
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
X-Asterisk-HangupCause: Unknown
X-Asterisk-HangupCauseCode: 0
Content-Length: 0

Retransmitting #1 (NAT) to xxx.yyy.zzz.90:5060:
BYE sip:from at 192.168.128.205 SIP/2.0
Via: SIP/2.0/UDP xxx.yyy.zzz.94:5060;branch=z9hG4bK1890cc40;rport
Route: <sip:xxx.yyy.zzz.90;lr;ftag=dwage;did=643.3986f963>
From: <sip:to at xxx.yyy.zzz.90>;tag=as671f3653
To: "aaa" <sip:from at xxx.yyy.zzz.90>;tag=dwage
Call-ID: vpdjrkrfbsmjkvy at aalexandrov.site
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
X-Asterisk-HangupCause: Unknown
X-Asterisk-HangupCauseCode: 0
Content-Length: 0