[Kamailio-Users] No audio both ways.

Daniel-Constantin Mierla miconda at gmail.com
Fri Oct 10 09:17:53 CEST 2008


If all calls go like this, usage of a rtp relay (rttpproxy/mediaproxy) 
is no longer necessary -- I see in the config you call use_media_proxy() 
- asterisk will handle media relay if it has public ip if the nat option 
is enabled in asterisk as David says.

Cheers,
Daniel


On 10/09/08 12:07, David Villasmil wrote:
> Try adding a:
>
> nat=yes
>
> to the kamailio/openser peer definition and test
>
> dvg
>
> On Thu, Oct 9, 2008 at 10:58 AM, luzango mfupe 
> <luzango.mfupe at gmail.com <mailto:luzango.mfupe at gmail.com>> wrote:
>
>
>     Hi mates,
>     I have this setup:
>     Xlite---->Openser---->Asterisk------>VoIP to PTSN Provider
>
>     I use kamailio 1.3.3(port5060), MediaProxy and Asterisk 1.4 (port
>     5065) on the same Debian Box in Realtime with no NAT. Asterisk
>     connects calls to the VoIP to PSTN provider. I am able to
>     establish calls towards the PSTN side(Landline & Mobiles)  but
>     with no audio. I can hear the ringing tone but when the call
>     connects and the conversation begin i hear nothing so as the
>     Callee side. 
>
>     Below are my configs,the ngrep captured packets and codecs.
>     ####################################################################################
>     route[4] {
>         # routing to the public network
>         rewritehostport("xx.xxx.xxx.xx:5065");
>         t_on_failure("2");
>         if (!t_relay()) {
>             sl_reply_error();
>         };
>     exit;
>     }
>
>     route[6] {
>         #
>         # -- NAT handling --
>         #
>         if (isbflagset(6) || isbflagset(7)) {
>             append_hf("P-hint: Route[6]: mediaproxy \r\n");
>             use_media_proxy();
>         };
>     }
>
>     route[10] {
>         #from an internal domain -> inbound
>         #Native SIP destinations are handled using the location table
>         #Gateway destinations are handled by regular expressions
>         append_hf("P-hint: inbound->inbound \r\n");
>
>         if (uri=~"^sip:0[1-9][0-9]+ at .*") {
>             if (is_user_in("credentials","local")) {
>             strip(1);
>             prefix("27");
>                route(6);
>                route(4);
>                 exit;
>             } else {
>                 sl_send_reply("403", "No permissions for local calls");
>                 exit;
>             };
>         };
>
>     if (uri=~"^sip:00[1-9][0-9]+ at .*") {
>             if (is_user_in("credentials","int")) {
>                strip(2);
>                route(6);
>                 route(4);
>                 exit;
>             } else {
>                 sl_send_reply("403", "No permissions for international
>     calls");
>             };
>         };
>
>     ###################################################################################
>     This call was from the xlite softphone 1974 towards the Landline
>     0123825710.
>     ###################################################################################
>     U 2008/12/06 03:38:43.896057 196.212.209.18:46738
>     <http://196.212.209.18:46738> -> kamailio IP:5060
>       INVITE sip:0123825710 at KAMAILIO ip SIP/2.0..Via: SIP/2.0/UDP
>     192.168.0.55 <http://192.168.0.55>:
>       46738;branch=z9hG4bK-d8754z-2d5e73428b95d262-1---d8754z-;rport..Max-Forward
>       s: 70..Contact: <sip:1974 at 196.212.209.18:46738
>     <http://sip:1974@196.212.209.18:46738>>..To: "0123825710"<sip:01238
>       25710 at kamailio IP>..From: <sip:1974 at kamailio
>     IP>;tag=353dd217..Call-ID:
>        MGRiNzM0ZGYxZTk1ZDI3ZjZmMTRhMDVlNTQ0NTUwZjg...CSeq: 1
>     INVITE..Allow: INVIT
>       E, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
>     INFO..Cont
>       ent-Type: application/sdp..User-Agent: X-Lite release 1014k
>     stamp 47051..Co
>       ntent-Length: 237....v=0..o=- 1 2 IN IP4
>     192.168.0.55..s=CounterPath X-Lite
>        3.0..c=IN IP4 192.168.0.55..t=0 0..m=audio 60782 RTP/AVP 0 8 3
>     101..a=fmtp
>       :101 0-15..a=rtpmap:101 telephone-event/8000..a=alt:1 1 :
>     ljiQYRpD NiMZsfdZ
>        192.168.0.55 <http://192.168.0.55> 60782..a=sendrecv..
>
>     U 2008/12/06 03:38:43.896350 Kamailio IP:5060 ->
>     196.212.209.18:46738 <http://196.212.209.18:46738>
>       SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP
>     192.168.0.55:46 <http://192.168.0.55:46>
>       738;branch=z9hG4bK-d8754z-2d5e73428b95d262-1---d8754z-;rport=46738;received
>       =196.212.209.18..To: "0123825710"<sip:0123825710 at kamailio
>     IP>;tag=329cfea
>       a6ded039da25ff8cbb8668bd2.dcfe..From: <sip:1974 at kamailio
>     IP>;tag=353dd217
>       ..Call-ID: MGRiNzM0ZGYxZTk1ZDI3ZjZmMTRhMDVlNTQ0NTUwZjg...CSeq: 1
>     INVITE..Pr
>       oxy-Authenticate: Digest realm="41.208.212.97
>     <http://41.208.212.97>", nonce="4939d8cfeb060ab14354
>       85eee811cdf644f759a2"..Content-Length: 0....
>
>     U 2008/12/06 03:38:44.086313 196.212.209.18:46738
>     <http://196.212.209.18:46738> -> kamailio IP:5060
>       ACK sip:0123825710 at kamailio IP SIP/2.0..Via: SIP/2.0/UDP
>     192.168.0.55:467 <http://192.168.0.55:467>
>       38;branch=z9hG4bK-d8754z-2d5e73428b95d262-1---d8754z-;rport..To:
>     "012382571
>       0"<sip:0123825710 at kamailio
>     IP>;tag=329cfeaa6ded039da25ff8cbb8668bd2.dcfe.
>       .From: <sip:1974 at 41.208.212.97
>     <mailto:sip%3A1974 at 41.208.212.97>>;tag=353dd217..Call-ID:
>     MGRiNzM0ZGYxZTk1ZDI3
>       ZjZmMTRhMDVlNTQ0NTUwZjg...CSeq: 1 ACK..Content-Length: 0....
>
>
>
>     U 2008/12/06 03:38:44.582208 asterisk IP:5065 -> kamailio IP:5060
>       SIP/2.0 100 Trying..Via: SIP/2.0/UDP kamailio
>     IP;branch=z9hG4bKd78.7bd576
>       24.0;received=kamailio IP..Via: SIP/2.0/UDP
>     192.168.0.55:46738;received=1
>       96.212.209.18
>     <http://96.212.209.18>;branch=z9hG4bK-d8754z-2aedec27a8ddd96c-1---d8754z-;rport=4673
>       8..Record-Route: <sip:kamailio
>     IP;lr=on;ftag=353dd217;nat=yes>..From: <si
>       p:1974 at kamailio IP>;tag=353dd217..To:
>     "0123825710"<sip:0123825710 at 41.208.
>       212.97>..Call-ID:
>     MGRiNzM0ZGYxZTk1ZDI3ZjZmMTRhMDVlNTQ0NTUwZjg...CSeq: 2 INV
>       ITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL,
>     OPTIONS, BYE, RE
>       FER, SUBSCRIBE, NOTIFY..Supported: replaces..Contact:
>     <sip:27123825710 at 41.2
>       08.212.97:5065>..Content-Length: 0....
>
>     U 2008/12/06 03:38:44.711225 70.42.72.49:5060
>     <http://70.42.72.49:5060> -> Asterisk IP:5065
>       SIP/2.0 100 Giving a try..Via: SIP/2.0/UDP Asterisk
>     IP:5065;branch=z9hG4b
>       K22abec12;rport=5065..From: "1974" <sip:1974 at Asterisk
>     IP:5065>;tag=as6a7c
>       b89f..To: <sip:1214650027123825710 at 70.42.72.49
>     <mailto:sip%3A1214650027123825710 at 70.42.72.49>>..Call-ID:
>     1934f5d443abffe07
>       c59d0a42215b49c at 41.208.212.97..CSeq: 102 INVITE..Server: OpenSER
>     (1.3.2-not
>       ls (i386/solaris))..Content-Length: 0....
>
>     U 2008/12/06 03:38:47.206445 70.42.72.49:5060
>     <http://70.42.72.49:5060> -> Asterisk IP:5065
>       SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP Asterisk
>     IP:5065;branch=z9
>       hG4bK22abec12;rport=5065..From: "1974" <sip:1974 at Asterisk
>     IP:5065>;tag=as
>       6a7cb89f..To: <sip:1214650027123825710 at 70.42.72.49
>     <mailto:sip%3A1214650027123825710 at 70.42.72.49>>;tag=cba-1a6e-48ecbab6..
>       Call-ID: 1934f5d443abffe07c59d0a42215b49c at Asterisk IP..CSeq: 102
>     INVITE..
>       Contact: <sip:1214650027123825710 at 70.42.72.138
>     <mailto:sip%3A1214650027123825710 at 70.42.72.138>>..Date: Wed, 08
>     Oct 2008 13:
>       50:49 GMT..Server: BRSIP v2.0.1.2..Record-Route:
>     <sip:70.42.72.49 <http://70.42.72.49>;lr=on;fta
>       g=as6a7cb89f>..Content-Type: application/sdp..Content-Length:
>     212....v=0..o
>       =BRSDP 792898 792898 IN IP4 216.49.201.22..s=BRSDP Session..c=IN
>     IP4 216.49
>       .201.22..t=0 0..m=audio 27852 RTP/AVP 0
>     101..a=ptime:20..a=rtpmap:0 PCMU/80
>       00..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..
>
>     U 2008/12/06 03:38:47.206891 Asterisk IP:5065 -> Kamailio IP:5060
>       SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP kamailio
>     IP;branch=z9hG4bK
>       d78.7bd57624.0;received=kamailio IP..Via: SIP/2.0/UDP
>     192.168.0.55:46738 <http://192.168.0.55:46738>;
>       received=196.212.209.18
>     <http://196.212.209.18>;branch=z9hG4bK-d8754z-2aedec27a8ddd96c-1---d8754z-;
>       rport=46738..Record-Route: <sip:41.208.212.97
>     <http://41.208.212.97>;lr=on;ftag=353dd217;nat=yes>.
>       .From: <sip:1974 at kamilio IP>;tag=353dd217..To:
>     "0123825710"<sip:01238257
>       10 at 41.208.212.97
>     <mailto:10 at 41.208.212.97>>;tag=as4377a96d..Call-ID:
>     MGRiNzM0ZGYxZTk1ZDI3ZjZmMTRhMDVl
>       NTQ0NTUwZjg...CSeq: 2 INVITE..User-Agent: Asterisk PBX..Allow:
>     INVITE, ACK,
>        CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Supported:
>     replaces..Conta
>       ct: <sip:27123825710 at Asterisk IP:5065>..Content-Type:
>     application/sdp..Co
>       ntent-Length: 287....v=0..o=root 2664 2664 IN IP4
>     41.208.212.97..s=session.
>      .c=IN IP4 41.208.212.97..t=0 0..m=audio 19202 RTP/AVP 8 0 3
>     101..a=rtpmap:8
>        PCMA/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:3
>     GSM/8000..a=rtpmap:101 telepho
>       ne-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - -
>     -..a=ptime:20..a=se
>       ndrecv..
>
>     U 2008/12/06 03:38:47.207106 Kamailio IP:5060 ->
>     196.212.209.18:46738 <http://196.212.209.18:46738>
>       SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP
>     192.168.0.55:46738;received=
>       196.212.209.18
>     <http://196.212.209.18>;branch=z9hG4bK-d8754z-2aedec27a8ddd96c-1---d8754z-;rport=467
>       38..Record-Route: <sip:kamailio
>     IP;lr=on;ftag=353dd217;nat=yes>..From: <s
>       ip:1974 at Kamailio IP>;tag=353dd217..To:
>     "0123825710"<sip:0123825710 at kamilio IP>;tag=as4377a96d..Call-ID:
>     MGRiNzM0ZGYxZTk1ZDI3ZjZmMTRhMDVlNTQ0NTUwZ
>       jg...CSeq: 2 INVITE..User-Agent: Asterisk PBX..Allow: INVITE,
>     ACK, CANCEL,
>       OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Supported:
>     replaces..Contact: <sip:
>       27123825710 at 41.208.212.97:5065
>     <http://27123825710@41.208.212.97:5065>>..Content-Type:
>     application/sdp..Content-Len
>       gth: 287....v=0..o=root 2664 2664 IN IP4 Asterisk
>     IP..s=session..c=IN IP4
>        41.208.212.97..t=0 0..m=audio 19202 RTP/AVP 8 0 3
>     101..a=rtpmap:8 PCMA/800
>       0..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a=rtpmap:101
>     telephone-event/
>       8000..a=fmtp:101 0-16..a=silenceSupp:off - - -
>     -..a=ptime:20..a=sendrecv..
>
>     ########################################################################################
>     Sip.conf
>     [general]
>     context=from-trunk                    
>     bindport=5065                   
>     autocreatepeer=yes                              
>     bindaddr=xx.xxx.xxx.xx                     
>                                     
>                                     
>     disallow=all                    
>     ;allow=gsm
>     ;allow=amr
>     allow=alaw
>     allow=ulaw                      
>     allow=gsm
>     ;allow=ilbc                     
>     ;disallow=all
>     ;
>     ;useragent=Asterisk PBX         
>     dtmfmode = rfc2833              
>                                                                       
>                                 
>     domain=xx.xxx.xxx.xx                   ; Add IP address as local
>     domain
>                                     
>     [Provider]
>     disallow=all
>     canreinvite=no
>     context=from-trunk
>     allow=all
>     ;allow=ulaw
>     ;allow=gsm
>     host=xx.xx.xx.xx
>     insecure=port,invite
>     type=peer                               ; we only want to call
>     out, not be call$
>     dtmfmode=rfc2833
>     #########################################################################################
>     Here is my codecs
>
>     41*CLI> core show translation
>              Translation times between formats (in milliseconds) for
>     one second of data
>               Source Format (Rows) Destination Format (Columns)
>
>               g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex
>     ilbc g726 g722 amr
>          g723    -   -    -    -        -     -    -     -    -     -
>        -    -    -   -
>           gsm    -   -    2    2        2     2    1     5    -    19
>        -    2    -  14
>          ulaw    -   5    -    1        2     2    1     5    -    19
>        -    2    -  14
>          alaw    -   5    1    -        2     2    1     5    -    19
>        -    2    -  14
>      g726aal2    -   5    2    2        -     2    1     5    -    19
>        -    1    -  14
>         adpcm    -   5    2    2        2     -    1     5    -    19
>        -    2    -  14
>          slin    -   4    1    1        1     1    -     4    -    18
>        -    1    -  13
>         lpc10    -   6    3    3        3     3    2     -    -    20
>        -    3    -  15
>          g729    -   -    -    -        -     -    -     -    -     -
>        -    -    -   -
>         speex    -   6    3    3        3     3    2     6    -     -
>        -    3    -  15
>          ilbc    -   -    -    -        -     -    -     -    -     -
>        -    -    -   -
>          g726    -   5    2    2        1     2    1     5    -    19
>        -    -    -  14
>          g722    -   -    -    -        -     -    -     -    -     -
>        -    -    -   -
>           amr    -   6    3    3        3     3    2     6    -    20
>        -    3    -   -
>      
>
>
>     With best regards,
>     Lu.
>
>     -- 
>     Luzango Mfupe
>     TUUNE MOBILE
>     Tel:0128440528/0123825710
>     Tshwane-RSA
>
>     "...Ships are safe in harbor, but they were never meant to stay
>     there......."
>
>     _______________________________________________
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>
>
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>
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>   

-- 
Daniel-Constantin Mierla
http://www.asipto.com





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