[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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