Re: how can opensips support both IPv4 and IPv6?

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Re: how can opensips support both IPv4 and IPv6?

Bogdan-Andrei Iancu
Hi Jack,

OpenSIPS deals with signalling also (via 2 interfaces), so obviously you
need to use rtpproxy (in bridging mode) to deal with the RTP part.

Regards,
Bogdan

Jack Miao wrote:

> Thank you a lot!
> But I prefer to use RTPproxy to implement it?
>
>
> On Mon, Nov 3, 2008 at 5:33 PM, Bogdan-Andrei Iancu
> <[hidden email] <mailto:[hidden email]>> wrote:
>
>     Hi Jack,
>
>     You have to configure 2 interfaces in opensips (2 listen
>     directives), one for IPv4 and one for IPv6.
>
>     Regards,
>     Bogdan
>
>     Jack Miao wrote:
>
>           Intercommunication of sip terminal between IPv4 and IPv6 is
>         really a problem.could someone show a example configuration
>         about this subject?Or tell me about the train of thought using
>         module function?
>          Thanks a lot!
>         ------------------------------------------------------------------------
>
>         _______________________________________________
>         Users mailing list
>         [hidden email] <mailto:[hidden email]>
>         http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>          
>
>
>


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Re: how can opensips support both IPv4 and IPv6?

troxlinux
Hi Bogdan, as I can have my rttproxy in bridge mode, you have an example?

in my server sip I have two net cards one it public and a private

greetings

rickygm

2008/11/6 Bogdan-Andrei Iancu <[hidden email]>
Hi Jack,

OpenSIPS deals with signalling also (via 2 interfaces), so obviously you
need to use rtpproxy (in bridging mode) to deal with the RTP part.




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Re: how can opensips support both IPv4 and IPv6?

Bogdan-Andrei Iancu
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

Regards,
Bogdan

troxlinux wrote:

> Hi Bogdan, as I can have my rttproxy in bridge mode, you have an example?
>
> in my server sip I have two net cards one it public and a private
>
> greetings
>
> rickygm
>
> 2008/11/6 Bogdan-Andrei Iancu <[hidden email]
> <mailto:[hidden email]>>
>
>     Hi Jack,
>
>     OpenSIPS deals with signalling also (via 2 interfaces), so
>     obviously you
>     need to use rtpproxy (in bridging mode) to deal with the RTP part.
>
>
>


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Re: how can opensips support both IPv4 and IPv6?

troxlinux
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@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: <[hidden email]>;tag=cfc7ce84fac4cb57
To: <[hidden email]>
Contact: <sip:119@192.168.10.28:5060;nat=yes;nat=yes>
Supported: replaces, timer, path
Call-ID: [hidden email]
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@192.168.10.19:5063 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bKb305.df34945.0
From: <[hidden email]>;tag=cfc7ce84fac4cb57
Call-ID: [hidden email]
To: <[hidden email]>
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: <[hidden email]>;tag=cfc7ce84fac4cb57
To: <[hidden email]>;tag=as3a67d233
Call-ID: [hidden email]
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@192.168.10.1:5070 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bKb305.df34945.1
From: <[hidden email]>;tag=cfc7ce84fac4cb57
Call-ID: [hidden email]
To: <[hidden email]>;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: <[hidden email]>;tag=cfc7ce84fac4cb57
To: <[hidden email]>;tag=as3a67d233
Call-ID: [hidden email]
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 [hidden email] SIP/2.0
Via: SIP/2.0/UDP 192.168.10.28:5060;branch=z9hG4bKc80584c1e2a47002
From: <[hidden email]>;tag=cfc7ce84fac4cb57
To: <[hidden email]>;tag=as3a67d233
Contact: <sip:119@192.168.10.28:5060>
Proxy-Authorization: Digest username="119", realm="192.168.10.1", algorithm=MD5, uri="[hidden email]", nonce="491d0af68c7aade3e86ae38c262008b5141d5769", response="804c1e89ed73b32be25f010495524aca"
Call-ID: [hidden email]
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: <[hidden email]>;tag=9c88b01629d86766i3
From: <[hidden email]>;tag=cfc7ce84fac4cb57
Call-ID: [hidden email]
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 <[hidden email]>
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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