[OpenSIPS-Users] how can opensips support both IPv4 and IPv6?
troxlinux
xserverlinux at gmail.com
Fri Nov 14 06:29:55 CET 2008
Hi Bogdan , I have my server sip integrated with asterisk for voicemail and
meetme, adds to rtpproxy to solve my problems of nat with my remote clients,
the result was almost satisfactory some details to improve, the problem was
that after adding rtpproxy when my clients do not answer a call no longer
here jumps me to the voicemail only shows in the 488 Not Aceptable here..
UAC == NAT == internet ===wan === eth0 server sip/asterisk === eth1 === UAC
clients
asterisk port : 5070
sip proxy: 5060
I have tried to put the force_rtp_proxy ("", 192.168.1.64); , but it doesn't
always work me on he writes the sdp twice
onreply_route[1] {
if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])"){
force_rtp_proxy();
# force_rtp_proxy("","192.168.1.64");
append_hf("P-hint: onreply_route|force_rtp_proxy \r\n");
}
if (!search("^Content-Length:[ ]*0")) {
# search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes');
if (isbflagset(6)) {
append_hf("P-hint: Onreply-route - fixcontact \r\n");
fix_nated_contact();
}
}
exit;
}
my sip log:
U +1.050293 192.168.10.1:5060 -> 192.168.10.1:5070
INVITE sip:u116 at 192.168.10.1:5070 SIP/2.0
Record-Route: <sip:192.168.10.1;lr=on;ftag=cfc7ce84fac4cb57>
Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bKb305.df34945.1
Via: SIP/2.0/UDP 192.168.10.28:5060;rport=5060;branch=z9hG4bKc80584c1e2a47002
From: <sip:119 at 192.168.10.1 <sip%3A119 at 192.168.10.1>>;tag=cfc7ce84fac4cb57
To: <sip:116 at 192.168.10.1 <sip%3A116 at 192.168.10.1>>
Contact: <sip:119 at 192.168.10.28:5060;nat=yes;nat=yes>
Supported: replaces, timer, path
Call-ID: cac52e44f0f25722 at 192.168.10.28
CSeq: 3157 INVITE
User-Agent: Grandstream GXV3000 1.1.3.14
Max-Forwards: 69
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Type: application/sdp
Content-Length: 604
P-hint: inbound->inbound
P-hint: Route[20]: Rtpproxy
P-hint: Route[20]: Rtpproxy
v=0
o=119 8000 8001 IN IP4 192.168.10.28
s=SIP Call
*c=IN IP4 192.168.10.1192.168.10.1 *
t=0 0
m=audio 3505635056 RTP/AVP 18 4 3 2 0 101
a=sendrecv
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
m=video 3505835058 RTP/AVP 99 34
a=sendrecv
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=428014; packetization-mode=0;
sprop-parameter-sets=Z0KADJWgUH5A,aM48gM==
a=rtpmap:34 H263/90000
a=fmtp:34 CIF=2 MaxBR=1280
a=framerate:20
a=nortpproxy:yes
a=nortpproxy:yes
#
U +0.000051 192.168.10.1:5060 -> 192.168.10.19:5063
CANCEL sip:116 at 192.168.10.19:5063 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bKb305.df34945.0
From: <sip:119 at 192.168.10.1 <sip%3A119 at 192.168.10.1>>;tag=cfc7ce84fac4cb57
Call-ID: cac52e44f0f25722 at 192.168.10.28
To: <sip:116 at 192.168.10.1 <sip%3A116 at 192.168.10.1>>
CSeq: 3157 CANCEL
Max-Forwards: 70
Content-Length: 0
#
U +0.000519 192.168.10.1:5070 -> 192.168.10.1:5060
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP
192.168.10.1;branch=z9hG4bKb305.df34945.1;received=192.168.10.1
Via: SIP/2.0/UDP 192.168.10.28:5060;rport=5060;branch=z9hG4bKc80584c1e2a47002
From: <sip:119 at 192.168.10.1 <sip%3A119 at 192.168.10.1>>;tag=cfc7ce84fac4cb57
To: <sip:116 at 192.168.10.1 <sip%3A116 at 192.168.10.1>>;tag=as3a67d233
Call-ID: cac52e44f0f25722 at 192.168.10.28
CSeq: 3157 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
#
U +0.000142 192.168.10.1:5060 -> 192.168.10.1:5070
ACK sip:u116 at 192.168.10.1:5070 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bKb305.df34945.1
From: <sip:119 at 192.168.10.1 <sip%3A119 at 192.168.10.1>>;tag=cfc7ce84fac4cb57
Call-ID: cac52e44f0f25722 at 192.168.10.28
To: <sip:116 at 192.168.10.1 <sip%3A116 at 192.168.10.1>>;tag=as3a67d233
CSeq: 3157 ACK
Max-Forwards: 70
Content-Length: 0
#
U +0.000174 192.168.10.1:5060 -> 192.168.10.28:5060
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 192.168.10.28:5060;rport=5060;branch=z9hG4bKc80584c1e2a47002
From: <sip:119 at 192.168.10.1 <sip%3A119 at 192.168.10.1>>;tag=cfc7ce84fac4cb57
To: <sip:116 at 192.168.10.1 <sip%3A116 at 192.168.10.1>>;tag=as3a67d233
Call-ID: cac52e44f0f25722 at 192.168.10.28
CSeq: 3157 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
#
U +0.002025 192.168.10.28:5060 -> 192.168.10.1:5060
ACK sip:116 at 192.168.10.1 <sip%3A116 at 192.168.10.1> SIP/2.0
Via: SIP/2.0/UDP 192.168.10.28:5060;branch=z9hG4bKc80584c1e2a47002
From: <sip:119 at 192.168.10.1 <sip%3A119 at 192.168.10.1>>;tag=cfc7ce84fac4cb57
To: <sip:116 at 192.168.10.1 <sip%3A116 at 192.168.10.1>>;tag=as3a67d233
Contact: <sip:119 at 192.168.10.28:5060>
Proxy-Authorization: Digest username="119", realm="192.168.10.1",
algorithm=MD5, uri="sip:116 at 192.168.10.1 <sip%3A116 at 192.168.10.1>",
nonce="491d0af68c7aade3e86ae38c262008b5141d5769",
response="804c1e89ed73b32be25f010495524aca"
Call-ID: cac52e44f0f25722 at 192.168.10.28
CSeq: 3157 ACK
User-Agent: Grandstream GXV3000 1.1.3.14
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0
#
U +0.006401 192.168.10.19:5063 -> 192.168.10.1:5060
SIP/2.0 487 Request Terminated
To: <sip:116 at 192.168.10.1 <sip%3A116 at 192.168.10.1>>;tag=9c88b01629d86766i3
From: <sip:119 at 192.168.10.1 <sip%3A119 at 192.168.10.1>>;tag=cfc7ce84fac4cb57
Call-ID: cac52e44f0f25722 at 192.168.10.28
CSeq: 3157 INVITE
Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bKb305.df34945.0
Via: SIP/2.0/UDP 192.168.10.28:5060;rport=5060;branch=z9hG4bKc80584c1e2a47002
Record-Route: <sip:192.168.10.1;lr=on;ftag=cfc7ce84fac4cb57>
Server: Linksys/SPA942-6.1.3(a)
Content-Length: 0
Regards
rickygm
2008/11/13 Bogdan-Andrei Iancu <bogdan at voice-system.ro>
> See the help from rtpproxy:
>
> usage: rtpproxy [-2fv] [-l addr1[/addr2]] [-6 addr1[/addr2]] [-s path]
> [-t tos] [-r rdir [-S sdir]] [-T ttl] [-L nfiles]
>
> and the manual for nathelper (flags for force_rtp_proxy):
> http://www.opensips.org/html/docs/modules/1.4.x/nathelper.html#id2515879
>
>
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