[SR-Users] force_rtp_proxy/rtpproxy_offer/answer not replacing ip addr.

MingHon gminghon at gmail.com
Wed Jul 6 10:33:52 CEST 2011


Hi,

Thanks for your reply..

RTPProxy and kamailio is running on the same centos box.

below is the command how i connect both RTPProxy and Kamailio

/----Kamailio----/

#!ifdef WITH_NAT
modparam("rtpproxy", "rtpproxy_sock", "udp:localhost:7722")
#!endif

/----RTPProxy----/

rtpproxy -l 192.168.1.3 -s udp:*:7722 -m 10000 -M 20000 -u user

/----kamctl fifo nh_show_rtpp----/

udp:localhost:7722::  set=0
index:: 0
disabled:: 0
weight:: 1
recheck_ticks:: 0

/--------/

im using kamailio ver. 3.1.4 and rtpproxy ver. 1.2.1


Please advice..

Thank you very much for your help.



On Wed, Jul 6, 2011 at 4:16 PM, Carsten Bock <carsten at ng-voice.com> wrote:

> Hi,
>
> Note:
> The methods of rtpproxy-module will only replace the IP, if the
> Kamailio can access the RTPProxy.
>
> How is your RTPProxy connected to your Kamailio? Socket or TCP? Do you
> have the Kamailio FIFO enabeld?
> If you have the fifo enabled, you should check the following:
>
> kamctl fifo nh_show_rtpp
>
> You should see, that the Kamailio is connected to the RTPProxy. If no,
> then that is your problem.
> If the RTPProxy is connected and is listening on the TCP socket, then
> you can do an ngrep to see the communication between Kamailio and
> RTPProxy, which might help you further with your investigation.
>
> Carsten
>
> 2011/7/6 MingHon <gminghon at gmail.com>:
> > Hi Carsten,
> > no is not about just rewriting the SDP.
> > i need my UACs media to relay on my rtpproxy
> > currently my UACs are sending the media to a private ip.
> > my rtpproxy is in behind nat and UACs behind another nat.
> >
> > On Wed, Jul 6, 2011 at 3:15 PM, Carsten Bock <carsten at ng-voice.com>
> wrote:
> >>
> >> Hi MingHon,
> >>
> >> what do you want to achieve? If it is only about rewritibng the SDP,
> >> then this will help you:
> >>
> >> fix_nated_sdp("10", "<your-ip-here>");
> >> => 0x02 rewrite media IP address (c=) with the provided IP address
> >> => 0x08 rewrite IP from origin description (o=) with the provided IP
> >> address
> >>
> >> Kind regards,
> >> Carsten
> >>
> >> 2011/7/6 MingHon <gminghon at gmail.com>:
> >> > hello List,
> >> > anyone could give some hints??
> >> > im still unable to rewrite the sdp body.
> >> > hope to hear from you all.
> >> > thanks
> >> > --
> >> > Regards,
> >> >
> >> > MingHon
> >> >
> >> >
> >> > On Tue, Jul 5, 2011 at 3:49 PM, MingHon <gminghon at gmail.com> wrote:
> >> >>
> >> >> Hi List,
> >> >> im facing an issue that my kamailio proxy did not replace the ip
> >> >> address
> >> >> in the invite and 200OK sdp body.
> >> >> my rtpproxy is running: rtpproxy -l 192.168.1.3 -u:*:7722 -u user
> >> >> my kamailio is listening on 192.168.1.3, also
> >> >> define: advertised_address="175.136.223.112"; & advertised_port=5060;
> >> >> and my asterisk is on 192.168.1.23.
> >> >> sip signalling and rtp port forwarded to kamailio.
> >> >> uacs from another nat register successfully.
> >> >> if i put 2 lines of force_rtp_proxy("fcow","175.136.223.112");
> >> >> i will get double ip addr in c and o but kamailio ignore my ip addr.
> >> >> example i will get
> >> >> c=IN IP4 192.168.1.3192.168.1.3
> >> >> here is part of my simple script.
> >> >> hope you can help.
> >> >> thank you very much.
> >> >> ---------------cfg-------------------
> >> >> route[RTPPROXY] {
> >> >> #!ifdef WITH_NAT
> >> >> if (is_method("BYE")) {
> >> >> unforce_rtp_proxy();
> >> >> } else if (is_method("INVITE")){
> >> >> force_rtp_proxy("fcow","175.136.223.112");
> >> >> #force_rtp_proxy("fcow","175.136.223.112");
> >> >> xlog("L_INFO","offer");
> >> >> }
> >> >> if (!has_totag()) add_rr_param(";nat=yes");
> >> >> #!endif
> >> >> return;
> >> >> }
> >> >> --------------------------------------
> >> >> and here is the wireshark for uac INVITE and OK.
> >> >> -----------INVITE-----------------
> >> >> ve0
> >> >> EE;p9INVITE sip:102 at 192.168.2.132:5062 SIP/2.0
> >> >> Record-Route: <sip:192.168.1.3;lr=on;ftag=as032358a3;nat=yes>
> >> >> Via: SIP/2.0/UDP 192.168.1.3;branch=z9hG4bK09d5.c5e9e8d2.0
> >> >> Via: SIP/2.0/UDP 192.168.1.23:5080;branch=z9hG4bK71c27189;rport=5080
> >> >> Max-Forwards: 69
> >> >> From: "101" <sip:102 at aextddns.dyndns.info>;tag=as032358a3
> >> >> To: <sip:102 at 192.168.1.3:5060>
> >> >> Contact: <sip:102 at 192.168.1.23:5080>
> >> >> Call-ID: 416f6e09674ae9671bb7144a1cb11137 at aextddns.dyndns.info
> >> >> CSeq: 102 INVITE
> >> >> User-Agent: Asterisk PBX 1.6.2.18
> >> >> Date: Tue, 05 Jul 2011 07:20:53 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 1639709788 1639709788 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 10072 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
> >> >> -----------200OK---------------
> >> >> e90
> >> >> ElE;pX4tSIP/2.0 200 OK
> >> >> Via: SIP/2.0/UDP
> >> >>
> >> >> 192.168.2.200:5062
> ;rport=2788;received=175.138.21.31;branch=z9hG4bK2086380416
> >> >> Record-Route: <sip:192.168.1.3;lr=on;ftag=1796959074;nat=yes>
> >> >> From: "101" <sip:101 at aextddns.dyndns.info>;tag=1796959074
> >> >> To: <sip:102 at aextddns.dyndns.info>;tag=as2e4c0125
> >> >> Call-ID: 1985782590 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: 286
> >> >> v=0
> >> >> o=root 403900934 403900934 IN IP4 192.168.1.23
> >> >> s=Asterisk PBX 1.6.2.18
> >> >> c=IN IP4 192.168.1.23
> >> >> t=0 0
> >> >> m=audio 14420 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
> >> >> ------------------------------------
> >> >> My kamailio log.
> >> >> -----------LOG------------------
> >> >> DEBUG: rtpproxy [rtpproxy_funcs.c:148]: type <application/sdp> found
> >> >> valid
> >> >> DEBUG: rtpproxy [rtpproxy.c:2188]: proxy reply: 10070 192.168.1.3
> >> >> INFO: <script>: offer
> >> >> -------------------------------------
> >> >> double force_rtp_proxy
> >> >> --------kamailio -> asterisk [INVITE]---------
> >> >> Pyi-}E7V@:#pINVITE sip:102 at aextddns.dyndns.info SIP/2.0
> >> >> Record-Route: <sip:192.168.1.3;lr=on;ftag=640933430;nat=yes>
> >> >> Via: SIP/2.0/UDP 192.168.1.3;branch=z9hG4bK89a5.53e9f766.0
> >> >> Via: SIP/2.0/UDP
> >> >>
> >> >> 192.168.2.200:5062
> ;rport=2788;received=175.138.21.31;branch=z9hG4bK1673765648
> >> >> From: "101" <sip:101 at aextddns.dyndns.info>;tag=640933430
> >> >> To: <sip:102 at aextddns.dyndns.info>
> >> >> Call-ID: 1909950509 at 192.168.2.200
> >> >> CSeq: 21 INVITE
> >> >> Contact: <sip:101 at 175.138.21.31:2788>
> >> >> Content-Type: application/sdp
> >> >> Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY,
> >> >> REGISTER,
> >> >> SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
> >> >> Max-Forwards: 69
> >> >> User-Agent: T20 9.41.0.80
> >> >> Allow-Events: talk,hold,conference,refer,check-sync
> >> >> Content-Length: 334
> >> >> v=0
> >> >> o=20073 20073 IN IP4 192.168.1.3192.168.1.3
> >> >> s=SDP data
> >> >> c=IN IP4 192.168.1.3192.168.1.3
> >> >> t=0 0
> >> >> m=audio 1006410064 RTP/AVP 0 8 18 9 101
> >> >> a=rtpmap:0 PCMU/8000
> >> >> a=rtpmap:8 PCMA/8000
> >> >> a=rtpmap:18 G729/8000
> >> >> a=rtpmap:9 G722/8000
> >> >> a=fmtp:101 0-15
> >> >> a=rtpmap:101 telephone-event/8000
> >> >> a=sendrecv
> >> >> a=nortpproxy:yes
> >> >> a=nortpproxy:yes
> >> >> -----------LOG------------------
> >> >> DEBUG: rtpproxy [rtpproxy_funcs.c:148]: type <application/sdp> found
> >> >> valid
> >> >> DEBUG: rtpproxy [rtpproxy.c:2188]: proxy reply: 10068 192.168.1.3
> >> >> DEBUG: rtpproxy [rtpproxy_funcs.c:148]: type <application/sdp> found
> >> >> valid
> >> >> DEBUG: rtpproxy [rtpproxy.c:2188]: proxy reply: 10068 192.168.1.3
> >> >> INFO: <script>: offer
> >> >> -----------LOG------------------
> >> >>
> >> >> --
> >> >> Regards,
> >> >>
> >> >> MingHon
> >>
> >>
> >>
> >> --
> >> Carsten Bock
> >> http://www.ng-voice.com
> >> mailto:carsten at ng-voice.com
> >>
> >> Schomburgstr. 80
> >> 22767 Hamburg
> >> Germany
> >>
> >> Mobile +49 179 2021244
> >> Office +49 40 34927219
> >> Fax +49 40 34927220
> >
> >
> >
> > --
> > Regards,
> >
> > MingHon
> >
>
>
>
> --
> Carsten Bock
> http://www.ng-voice.com
> mailto:carsten at ng-voice.com
>
> Schomburgstr. 80
> 22767 Hamburg
> Germany
>
> Mobile +49 179 2021244
> Office +49 40 34927219
> Fax +49 40 34927220
>
> ~~~~~
> Checkout SIP-Provider CE:
> http://www.sipwise.com/products/spce/overview/
>



-- 
Regards,

MingHon
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