3 Feb
2005
3 Feb
'05
3:05 p.m.
I am not sure if I understand the question. If you want to redirect the
INVITE, then you need to get the new destination from somewhere --
either configure it statically in the configuration as you have done, or
use the user location database.
To use the user location database, call lookup("location") and then 3xx
back using sl_send reply:
lookup("location");
sl_send_reply("301", "Redirect");
That way SER would put the real destinations (registered by the called
user agent) into Contact and send it back to the calling user agent.
Jan.
On 03-02 15:14, George Lee wrote:
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@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@192.168.0.252;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.233;branch=z9hG4bKb537ec0f1387845f
From: "60000253"
<sip:60000253@192.168.0.252;user=phone>;tag=c526ecce44ec6d75
To: <sip:8000000@192.168.0.252;user=phone>
Contact: <sip:60000253@192.168.0.233;user=phone>
Supported: replaces
Call-ID: d6169571fe4f59a2(a)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@192.168.0.252;user=phone>;tag=c526ecce44ec6d75
To:
<sip:8000000@192.168.0.252;user=phone>;tag=b27e1a1d33761e85846fc98f5f3a7e58.f9ed
Call-ID: d6169571fe4f59a2(a)192.168.0.233
CSeq: 50105 INVITE
Contact: sip:80000000@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@192.168.0.252;user=phone
out_uri=sip:80000000@192.168.0.191:5060 via_cnt==1"
F4:
INVITE sip:80000000@192.168.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.233;branch=z9hG4bKc6905874d1906c58
From: "60000253"
<sip:60000253@192.168.0.252;user=phone>;tag=1be4a5d7d169d8f1
To: <sip:80000000@192.168.0.191:5060>
Contact: <sip:60000253@192.168.0.233;user=phone>
Supported: replaces
Call-ID: 0d325b0e778321ee(a)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@192.168.0.252;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.233;branch=z9hG4bKe966424e91ceac8e
From: "60000253"
<sip:60000253@192.168.0.252;user=phone>;tag=466d9b06b51ca7b4
To: <sip:80000002@192.168.0.252;user=phone>
Contact: <sip:60000253@192.168.0.233;user=phone>
Supported: replaces
Call-ID: c755c6b0908a8ce9(a)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@192.168.0.252;user=phone>;tag=466d9b06b51ca7b4
To:
<sip:80000002@192.168.0.252;user=phone>;tag=b27e1a1d33761e85846fc98f5f3a7e58.7906
Call-ID: c755c6b0908a8ce9(a)192.168.0.233
CSeq: 25248 INVITE
Contact: sip:80000000@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@192.168.0.252;user=phone
out_uri=sip:80000000@192.168.0.191:5060 via_cnt==1"
F8:
INVITE sip:80000000@192.168.0.191:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.233;branch=z9hG4bK475fff37270b762d
From: "60000253"
<sip:60000253@192.168.0.252;user=phone>;tag=58a90f4229e70a80
To: <sip:80000000@192.168.0.191:5060>
Contact: <sip:60000253@192.168.0.233;user=phone>
Supported: replaces
Call-ID: 2a79d9ac84e2f72e(a)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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