[Serusers] Dynamic Redirect Issue about SER!
George Lee
jlsnmp at 126.com
Thu Feb 3 08:14:13 CET 2005
SER gurus:
I am an newbie to SER usuage. At present a dynamic redirect issue happens to me. But it is very pity for SER not to support this case!!
Problem Description:
====================
When configuring SER as only redirect server, config file regarding redirect routing section is partly writen as follows:
#-----------------------request routing logic-------------------------
#main routing logic
if (method=="INVITE") {
#rewrite current URI, which is always part of destination ser
rewriteuri("sip:80000000 at 192.168.0.191:5060");
#redirect now
sl_send_reply("301", "Redirect");
break;
}
#---------------------------------------------------------------------
where, redirecting destination URI is staticly configed into SER, but dynamic destination URI is what our IP-PBX product(RTCCP) want to hope.
The following call flow diagram helps to understand above mentioned scenario.
GrandStream HandyTone486 SER Our B2BUA IP PBX(RTCCP)
IP: 192.168.0.253 IP: 192.168.0.252 IP: 192.168.0.191
+++++++++++++++++++++++++++++++++++++ First Call+++++++++++++++++++++++++++++++++++++++++++++++++
|-----------------F1(INVITE)---------->|
|<-----------------F2(302)------------>|
|-----------------F3(ACK)------------->|
|------------------------------------F4(INVITE)------------------------>|
( The subsequent call flow is omitted )
+++++++++++++++++++++++++++++++++++++ Second Call+++++++++++++++++++++++++++++++++++++++++++++++++
|-----------------F5(INVITE)---------->|
|<-----------------F6(302)------------>|
|-----------------F7(ACK)------------->|
|------------------------------------F8(INVITE)------------------------>|
( The subsequent call flow is omitted )
F1:
INVITE sip:8000000 at 192.168.0.252;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.233;branch=z9hG4bKb537ec0f1387845f
From: "60000253" <sip:60000253 at 192.168.0.252;user=phone>;tag=c526ecce44ec6d75
To: <sip:8000000 at 192.168.0.252;user=phone>
Contact: <sip:60000253 at 192.168.0.233;user=phone>
Supported: replaces
Call-ID: d6169571fe4f59a2 at 192.168.0.233
CSeq: 50105 INVITE
User-Agent: Grandstream HT487 1.0.5.16
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Content-Length: 330
v=0
o=60000253 0 8000 IN IP4 192.168.0.233
s=SIP Call
c=IN IP4 192.168.0.233
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 2 15 99
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:15 G728/8000
a=rtpmap:99 iLBC/8000
a=fmtp:99 mode=20
a=ptime:20
F2:
SIP/2.0 301 Redirect
Via: SIP/2.0/UDP 192.168.0.233;branch=z9hG4bKb537ec0f1387845f
From: "60000253" <sip:60000253 at 192.168.0.252;user=phone>;tag=c526ecce44ec6d75
To: <sip:8000000 at 192.168.0.252;user=phone>;tag=b27e1a1d33761e85846fc98f5f3a7e58.f9ed
Call-ID: d6169571fe4f59a2 at 192.168.0.233
CSeq: 50105 INVITE
Contact: sip:80000000 at 192.168.0.191:5060
Server: Sip EXpress router (0.8.12 (i386/linux))
Content-Length: 0
Warning: 392 192.168.0.252:5060 "Noisy feedback tells: pid=6155 req_src_ip=192.168.0.233 req_src_port=5060 in_uri=sip:8000000 at 192.168.0.252;user=phone out_uri=sip:80000000 at 192.168.0.191:5060 via_cnt==1"
F4:
INVITE sip:80000000 at 192.168.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.233;branch=z9hG4bKc6905874d1906c58
From: "60000253" <sip:60000253 at 192.168.0.252;user=phone>;tag=1be4a5d7d169d8f1
To: <sip:80000000 at 192.168.0.191:5060>
Contact: <sip:60000253 at 192.168.0.233;user=phone>
Supported: replaces
Call-ID: 0d325b0e778321ee at 192.168.0.233
CSeq: 17809 INVITE
User-Agent: Grandstream HT487 1.0.5.16
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Content-Length: 330
v=0
o=60000253 0 8000 IN IP4 192.168.0.233
s=SIP Call
c=IN IP4 192.168.0.233
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 2 15 99
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:15 G728/8000
a=rtpmap:99 iLBC/8000
a=fmtp:99 mode=20
a=ptime:20
F5:
INVITE sip:80000002 at 192.168.0.252;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.233;branch=z9hG4bKe966424e91ceac8e
From: "60000253" <sip:60000253 at 192.168.0.252;user=phone>;tag=466d9b06b51ca7b4
To: <sip:80000002 at 192.168.0.252;user=phone>
Contact: <sip:60000253 at 192.168.0.233;user=phone>
Supported: replaces
Call-ID: c755c6b0908a8ce9 at 192.168.0.233
CSeq: 25248 INVITE
User-Agent: Grandstream HT487 1.0.5.16
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Content-Length: 330
v=0
o=60000253 0 8000 IN IP4 192.168.0.233
s=SIP Call
c=IN IP4 192.168.0.233
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 2 15 99
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:15 G728/8000
a=rtpmap:99 iLBC/8000
a=fmtp:99 mode=20
a=ptime:20
F6:
SIP/2.0 301 Redirect
Via: SIP/2.0/UDP 192.168.0.233;branch=z9hG4bKe966424e91ceac8e
From: "60000253" <sip:60000253 at 192.168.0.252;user=phone>;tag=466d9b06b51ca7b4
To: <sip:80000002 at 192.168.0.252;user=phone>;tag=b27e1a1d33761e85846fc98f5f3a7e58.7906
Call-ID: c755c6b0908a8ce9 at 192.168.0.233
CSeq: 25248 INVITE
Contact: sip:80000000 at 192.168.0.191:5060
Server: Sip EXpress router (0.8.12 (i386/linux))
Content-Length: 0
Warning: 392 192.168.0.252:5060 "Noisy feedback tells: pid=6151 req_src_ip=192.168.0.233 req_src_port=5060 in_uri=sip:80000002 at 192.168.0.252;user=phone out_uri=sip:80000000 at 192.168.0.191:5060 via_cnt==1"
F8:
INVITE sip:80000000 at 192.168.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.233;branch=z9hG4bK475fff37270b762d
From: "60000253" <sip:60000253 at 192.168.0.252;user=phone>;tag=58a90f4229e70a80
To: <sip:80000000 at 192.168.0.191:5060>
Contact: <sip:60000253 at 192.168.0.233;user=phone>
Supported: replaces
Call-ID: 2a79d9ac84e2f72e at 192.168.0.233
CSeq: 62121 INVITE
User-Agent: Grandstream HT487 1.0.5.16
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Content-Length: 330
v=0
o=60000253 0 8000 IN IP4 192.168.0.233
s=SIP Call
c=IN IP4 192.168.0.233
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 2 15 99
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:15 G728/8000
a=rtpmap:99 iLBC/8000
a=fmtp:99 mode=20
a=ptime:20
In fact, I think SER should retrieve user portion of header "to" URI of Invite message(for above example it should be 80000000 for first call,whereas 80000002 for second call) to construct userinfo of destination redirect uri for header contact of 3xx response(but hostinfo maybe get from SER config file). Meanwhile I study into sl_send_reply source code, but no progress to issue.
So any suggestions or solutions are appreciated!!
Thanks in advance!!
George Lee(ShenZhen, CHINA)
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