[SR-Users] force_rtp_proxy[flag, ipaddress] not rewriting c and o ?

MingHon gminghon at gmail.com
Mon Jul 4 09:46:08 CEST 2011


Hello,

I also tried rtpproxy_offer/rtpproxy_answer but no luck.

my rtpproxy is running..

rtpproxy -l 192.168.2.3 -l 127.0.0.1:7722 -u user

and kamailio on 192.168.2.3 asterisk on 192.168.2.23

all three in behind same nat 175.136.223.112.

and uac is behind another nat.

below is my wireshark INVITE and 200OK.

route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) {
unforce_rtp_proxy(); } else if (is_method("INVITE")){
rtpproxy_offer("rco","175.136.223.112"); xlog("offer"); } if (!has_totag())
add_rr_param(";nat=yes"); #!endif

onreply_route[REPLY_ONE] {
        xdbg("incoming reply\n");
#!ifdef WITH_NAT
        if ((isflagset(FLT_NATS) || isbflagset(FLB_NATB))
                        && status=~"(183)|(2[0-9][0-9])") {
                rtpproxy_answer("rco","175.136.223.112");
                xlog("answer");
        }
        if (isbflagset("6")) {
                fix_nated_contact();
        }
#!endif

----------------------------------------------

e0 E!;pINVITE sip:102 at 192.168.2.132:5062 SIP/2.0 Record-Route:
<sip:192.168.1.3;nat=yes;ftag=as3d4c45ac;lr=on> Via: SIP/2.0/UDP
175.136.223.112:5060;branch=z9hG4bK7ed5.6ceec772.0 Via: SIP/2.0/UDP
192.168.1.23:5080;branch=z9hG4bK34b777a3;rport=5080 Max-Forwards: 69 From:
"101" <sip:102 at aextddns.dyndns.info>;tag=as3d4c45ac To: <
sip:102 at 192.168.1.3:5060> Contact: <sip:102 at 192.168.1.23:5080> Call-ID:
0bf35cba6b3cb98156b70a3a4db2507a at aextddns.dyndns.info CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.18 Date: Mon, 04 Jul 2011 07:28:08 GMT Allow:
INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported:
replaces, timer Content-Type: application/sdp Content-Length: 327 v=0 o=root
1789123892 1789123892 IN IP4 192.168.1.3 s=Asterisk PBX 1.6.2.18 c=IN IP4
192.168.1.3 t=0 0 m=audio 13260 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv
a=nortpproxy:yes

-----------------------------

e90 E|;YphSIP/2.0 200 OK Via: SIP/2.0/UDP
192.168.2.200:5062;rport=2474;received=175.138.21.31;branch=z9hG4bK823351926
Record-Route: <sip:192.168.1.3;nat=yes;ftag=1020708120;lr=on> From: "101" <
sip:101 at aextddns.dyndns.info>;tag=1020708120 To: <
sip:102 at aextddns.dyndns.info>;tag=as199c06be Call-ID:
987641369 at 192.168.2.200 CSeq: 21 INVITE Server: Asterisk PBX 1.6.2.18 Allow:
INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported:
replaces, timer Contact: <sip:102 at 192.168.1.23:5080> Content-Type:
application/sdp Content-Length: 304 v=0 o=root 1460646028 1460646028 IN IP4
192.168.1.3 s=Asterisk PBX 1.6.2.18 c=IN IP4 192.168.1.3 t=0 0 m=audio 18346
RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101
telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20
a=sendrecv a=nortpproxy:yes

-- 
Regards,

MingHon
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