g729 is not a free codec . YOu have to buy it from digium at rateof $10 per channel license . If you are just using asterisk and havent bought g729 license then asterisk will just do bridging of g729 and wont edit/transcode stream .
<br><br><div><span class="gmail_quote">On 16/11/06, <b class="gmail_sendername">Victor Toofic</b> <<a href="mailto:toofics@gmail.com">toofics@gmail.com</a>> wrote:</span><blockquote class="gmail_quote" style="margin-top: 0; margin-right: 0; margin-bottom: 0; margin-left: 0; margin-left: 0.80ex; border-left-color: #cccccc; border-left-width: 1px; border-left-style: solid; padding-left: 1ex">
I have the following scenario:<br><br> g729 gsm<br> UAS <-----------> * <-----------> UAC<br><br>I am using sipp to generate the calls between the UAC and the UAS and<br>sending some rtp from the UAC, I want * to do transcoding but as I see
<br>it is not. As long as I know 'Attempting native bridge' means only<br>passing-through the rtp ¿Am I wrong?<br><br>The UAC and UAS are registering with * properly:<br><br>--- sip.conf ------------------------------------------------------------
<br>[testgsm]<br>type=friend<br>host=dynamic<br>username=testgsm<br>context=astertest<br>canreinvite=no<br>disallow=all<br>allow=gsm<br><br>[testg729]<br>type=friend<br>host=dynamic<br>username=testg729<br>context=astertest
<br>canreinvite=no<br>disallow=all<br>allow=g729<br>-------------------------------------------------------------------------<br><br>-------------------------------------------------------------------------<br>dspam*CLI> sip show peers
<br>Name/username Host Dyn Nat ACL Port Status<br>testgsm/testgsm <a href="http://172.16.51.244">172.16.51.244</a> D 10000 Unmonitored<br>testg729/testg729 <a href="http://172.16.51.244">
172.16.51.244</a> D 20000 Unmonitored<br><br>-- Executing Answer("SIP/testgsm-081784b0", "") in new stack<br>-- Executing Wait("SIP/testgsm-081784b0", "1") in new stack<br>
-- Executing Dial("SIP/testgsm-081784b0", "SIP/testg729") in new stack<br>-- Called testg729<br>-- SIP/testg729-0817dd90 is ringing<br>-- SIP/testg729-0817dd90 answered SIP/testgsm-081784b0<br>-- Attempting native bridge of SIP/testgsm-081784b0 and SIP/testg729-0817dd90
<br>-------------------------------------------------------------------------<br><br>After the call is established the UAC is sending some RTP captured in a<br>pcap file in gsm:<br><br>-- tcpdump -T rtp udp ---------------------------------------------------
<br>15:58:31.868404 IP 172.16.51.244.10001 > 172.16.51.215.17050: udp/rtp 33 c3<br>15:58:31.868676 IP 172.16.51.215.15424 > 172.16.51.244.20001: udp/rtp 20 c18<br>15:58:31.895551 IP 172.16.51.244.10001 > 172.16.51.215.17050
: udp/rtp 33 c3<br>15:58:31.895775 IP 172.16.51.215.15424 > 172.16.51.244.20001: udp/rtp 20 c18<br>15:58:31.936468 IP 172.16.51.244.10001 > 172.16.51.215.17050: udp/rtp 33 c3<br>15:58:31.936477 IP 172.16.51.244.10001
> 172.16.51.215.17050: udp/rtp 33 c3<br>15:58:31.936711 IP 172.16.51.215.15424 > 172.16.51.244.20001: udp/rtp 20 c18<br>15:58:31.936908 IP 172.16.51.215.15424 > 172.16.51.244.20001: udp/rtp 20 c18<br>-------------------------------------------------------------------------
<br><br>Is there something wrong within the SDP? or Am I doing something wrong? Any<br>comments would be appreciated.. thanks!!<br><br>P.S. I am using Asterisk <a href="http://1.2.12.1">1.2.12.1</a> if that matters.<br><br>
--<br>Greetings...<br>Víctor Toofic<br><br><br><br>----------------------------------------------- 2006-11-15 16:15:12<br>UDP message sent:<br><br>INVITE sip:testg729@172.16.51.215:5060 SIP/2.0<br>Via: SIP/2.0/UDP <a href="http://172.16.51.244:34836">
172.16.51.244:34836</a>;branch=z9hG4bK-1-0<br>From: testgsm <sip:testgsm@172.16.51.215:5060>;tag=1<br>To: testg729 <sip:testg729@172.16.51.215:5060><br>Call-ID: <a href="mailto:1-28245@172.16.51.244">1-28245@172.16.51.244
</a><br>CSeq: 1 INVITE<br>Contact: sip:testgsm@172.16.51.244:34836<br>Max-Forwards: 70<br>Subject: Performance Test<br>Content-Type: application/sdp<br>Content-Length: 138<br><br>v=0<br>o=user1 53655765 2353687637 IN IP4
<a href="http://172.16.51.244">172.16.51.244</a><br>s=-<br>c=IN IP4 <a href="http://172.16.51.244">172.16.51.244</a><br>t=0 0<br>m=audio 10001 RTP/AVP 0<br>a=rtpmap:18 GSM/8000<br><br>----------------------------------------------- 2006-11-15 16:15:12
<br>UDP message received [404] bytes :<br><br>SIP/2.0 100 Trying<br>Via: SIP/2.0/UDP <a href="http://172.16.51.244:34836">172.16.51.244:34836</a>;branch=z9hG4bK-1-0;received=<a href="http://172.16.51.244">172.16.51.244</a>
<br>From: testgsm <sip:testgsm@172.16.51.215:5060>;tag=1<br>To: testg729 <sip:testg729@172.16.51.215:5060><br>Call-ID: <a href="mailto:1-28245@172.16.51.244">1-28245@172.16.51.244</a><br>CSeq: 1 INVITE<br>User-Agent: Asterisk PBX
<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Contact: <<a href="mailto:sip:testg729@172.16.51.215">sip:testg729@172.16.51.215</a>><br>Content-Length: 0<br><br>----------------------------------------------- 2006-11-15 16:15:12
<br>UDP message received [609] bytes :<br><br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP <a href="http://172.16.51.244:34836">172.16.51.244:34836</a>;branch=z9hG4bK-1-0;received=<a href="http://172.16.51.244">172.16.51.244</a><br>
From: testgsm <sip:testgsm@172.16.51.215:5060>;tag=1<br>To: testg729 <sip:testg729@172.16.51.215:5060>;tag=as14685910<br>Call-ID: <a href="mailto:1-28245@172.16.51.244">1-28245@172.16.51.244</a><br>CSeq: 1 INVITE
<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Contact: <<a href="mailto:sip:testg729@172.16.51.215">sip:testg729@172.16.51.215</a>><br>Content-Type: application/sdp
<br>Content-Length: 161<br><br>v=0<br>o=root 3567 3567 IN IP4 <a href="http://172.16.51.215">172.16.51.215</a><br>s=session<br>c=IN IP4 <a href="http://172.16.51.215">172.16.51.215</a><br>t=0 0<br>m=audio 17050 RTP/AVP 18
<br>a=rtpmap:18 GSM/8000<br>a=silenceSupp:off - - - -<br><br>----------------------------------------------- 2006-11-15 16:15:12<br>UDP message sent:<br><br>ACK sip:testg729@172.16.51.215:5060 SIP/2.0<br>Via: SIP/2.0/UDP
<a href="http://172.16.51.244:34836">172.16.51.244:34836</a>;branch=z9hG4bK-1-5<br>From: testgsm <sip:testgsm@172.16.51.215:5060>;tag=1<br>To: testg729 <sip:testg729@172.16.51.215:5060>;tag=as14685910<br>Call-ID:
<a href="mailto:1-28245@172.16.51.244">1-28245@172.16.51.244</a><br>CSeq: 1 ACK<br>Contact: sip:testgsm@172.16.51.244:34836<br>Max-Forwards: 70<br>Subject: Performance Test<br>Content-Length: 0<br><br>----------------------------------------------- 2006-11-15 16:16:15
<br>UDP message sent:<br><br>BYE sip:testg729@172.16.51.215:5060 SIP/2.0<br>Via: SIP/2.0/UDP <a href="http://172.16.51.244:34836">172.16.51.244:34836</a>;branch=z9hG4bK-1-9<br>From: testgsm <sip:testgsm@172.16.51.215:5060>;tag=1
<br>To: testg729 <sip:testg729@172.16.51.215:5060>;tag=as14685910<br>Call-ID: <a href="mailto:1-28245@172.16.51.244">1-28245@172.16.51.244</a><br>CSeq: 2 BYE<br>Contact: sip:testgsm@172.16.51.244:34836<br>Max-Forwards: 70
<br>Subject: Performance Test<br>Content-Length: 0<br><br>----------------------------------------------- 2006-11-15 16:16:15<br>UDP message received [453] bytes :<br><br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP <a href="http://172.16.51.244:34836">
172.16.51.244:34836</a>;branch=z9hG4bK-1-9;received=<a href="http://172.16.51.244">172.16.51.244</a><br>From: testgsm <sip:testgsm@172.16.51.215:5060>;tag=1<br>To: testg729 <sip:testg729@172.16.51.215:5060>;tag=as14685910
<br>Call-ID: <a href="mailto:1-28245@172.16.51.244">1-28245@172.16.51.244</a><br>CSeq: 2 BYE<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Contact: <<a href="mailto:sip:testg729@172.16.51.215">
sip:testg729@172.16.51.215</a>><br>Content-Length: 0<br>X-Asterisk-HangupCause: Normal Clearing<br><br><br><br>----------------------------------------------- 2006-11-15 16:15:14<br>UDP message received [770] bytes :<br>
<br>INVITE sip:testg729@172.16.51.244:20000 SIP/2.0<br>Via: SIP/2.0/UDP <a href="http://172.16.51.215:5060">172.16.51.215:5060</a>;branch=z9hG4bK5f760f5d;rport<br>From: "testgsm" <<a href="mailto:sip:testgsm@172.16.51.215">
sip:testgsm@172.16.51.215</a>>;tag=as23ee49d7<br>To: <sip:testg729@172.16.51.244:20000><br>Contact: <<a href="mailto:sip:testgsm@172.16.51.215">sip:testgsm@172.16.51.215</a>><br>Call-ID: <a href="mailto:375535bb4274d1ac67c51229526c3b8c@172.16.51.215">
375535bb4274d1ac67c51229526c3b8c@172.16.51.215</a><br>CSeq: 102 INVITE<br>User-Agent: Asterisk PBX<br>Max-Forwards: 70<br>Date: Wed, 15 Nov 2006 21:58:14 GMT<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
<br>Content-Type: application/sdp<br>Content-Length: 239<br><br>v=0<br>o=root 3567 3567 IN IP4 <a href="http://172.16.51.215">172.16.51.215</a><br>s=session<br>c=IN IP4 <a href="http://172.16.51.215">172.16.51.215</a><br>
t=0 0<br>m=audio 15424 RTP/AVP 18 101<br>a=rtpmap:18 G729/8000<br>a=fmtp:18 annexb=no<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=silenceSupp:off - - - -<br><br>----------------------------------------------- 2006-11-15 16:15:14
<br>UDP message sent:<br><br>SIP/2.0 180 Ringing<br>Via: SIP/2.0/UDP <a href="http://172.16.51.215:5060">172.16.51.215:5060</a>;branch=z9hG4bK5f760f5d;rport<br>From: "testgsm" <<a href="mailto:sip:testgsm@172.16.51.215">
sip:testgsm@172.16.51.215</a>>;tag=as23ee49d7<br>To: <sip:testg729@172.16.51.244:20000>;tag=1<br>Call-ID: <a href="mailto:375535bb4274d1ac67c51229526c3b8c@172.16.51.215">375535bb4274d1ac67c51229526c3b8c@172.16.51.215
</a><br>CSeq: 102 INVITE<br>Contact: <sip:<a href="http://172.16.51.244:20000">172.16.51.244:20000</a>;transport=UDP><br>Content-Length: 0<br><br>----------------------------------------------- 2006-11-15 16:15:14<br>
UDP message sent:<br><br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP <a href="http://172.16.51.215:5060">172.16.51.215:5060</a>;branch=z9hG4bK5f760f5d;rport<br>From: "testgsm" <<a href="mailto:sip:testgsm@172.16.51.215">
sip:testgsm@172.16.51.215</a>>;tag=as23ee49d7<br>To: <sip:testg729@172.16.51.244:20000>;tag=1<br>Call-ID: <a href="mailto:375535bb4274d1ac67c51229526c3b8c@172.16.51.215">375535bb4274d1ac67c51229526c3b8c@172.16.51.215
</a><br>CSeq: 102 INVITE<br>Contact: <sip:<a href="http://172.16.51.244:20000">172.16.51.244:20000</a>;transport=UDP><br>Content-Type: application/sdp<br>Content-Length: 139<br><br>v=0<br>o=user1 53655765 2353687637 IN IP4
<a href="http://172.16.51.244">172.16.51.244</a><br>s=-<br>c=IN IP4 <a href="http://172.16.51.244">172.16.51.244</a><br>t=0 0<br>m=audio 20001 RTP/AVP 0<br>a=rtpmap:18 G729/8000<br><br>----------------------------------------------- 2006-11-15 16:15:14
<br>UDP message received [398] bytes :<br><br>ACK sip:<a href="http://172.16.51.244:20000">172.16.51.244:20000</a>;transport=UDP SIP/2.0<br>Via: SIP/2.0/UDP <a href="http://172.16.51.215:5060">172.16.51.215:5060</a>;branch=z9hG4bK4fe64bf0;rport
<br>From: "testgsm" <<a href="mailto:sip:testgsm@172.16.51.215">sip:testgsm@172.16.51.215</a>>;tag=as23ee49d7<br>To: <sip:testg729@172.16.51.244:20000>;tag=1<br>Contact: <<a href="mailto:sip:testgsm@172.16.51.215">
sip:testgsm@172.16.51.215</a>><br>Call-ID: <a href="mailto:375535bb4274d1ac67c51229526c3b8c@172.16.51.215">375535bb4274d1ac67c51229526c3b8c@172.16.51.215</a><br>CSeq: 102 ACK<br>User-Agent: Asterisk PBX<br>Max-Forwards: 70
<br>Content-Length: 0<br><br>----------------------------------------------- 2006-11-15 16:16:15<br>UDP message received [398] bytes :<br><br>BYE sip:<a href="http://172.16.51.244:20000">172.16.51.244:20000</a>;transport=UDP SIP/2.0
<br>Via: SIP/2.0/UDP <a href="http://172.16.51.215:5060">172.16.51.215:5060</a>;branch=z9hG4bK59f6771e;rport<br>From: "testgsm" <<a href="mailto:sip:testgsm@172.16.51.215">sip:testgsm@172.16.51.215</a>>;tag=as23ee49d7
<br>To: <sip:testg729@172.16.51.244:20000>;tag=1<br>Contact: <<a href="mailto:sip:testgsm@172.16.51.215">sip:testgsm@172.16.51.215</a>><br>Call-ID: <a href="mailto:375535bb4274d1ac67c51229526c3b8c@172.16.51.215">
375535bb4274d1ac67c51229526c3b8c@172.16.51.215</a><br>CSeq: 103 BYE<br>User-Agent: Asterisk PBX<br>Max-Forwards: 70<br>Content-Length: 0<br><br>----------------------------------------------- 2006-11-15 16:16:15<br>UDP message sent:
<br><br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP <a href="http://172.16.51.215:5060">172.16.51.215:5060</a>;branch=z9hG4bK59f6771e;rport<br>From: "testgsm" <<a href="mailto:sip:testgsm@172.16.51.215">sip:testgsm@172.16.51.215
</a>>;tag=as23ee49d7<br>To: <sip:testg729@172.16.51.244:20000>;tag=1<br>Call-ID: <a href="mailto:375535bb4274d1ac67c51229526c3b8c@172.16.51.215">375535bb4274d1ac67c51229526c3b8c@172.16.51.215</a><br>CSeq: 103 BYE
<br>Contact: <sip:<a href="http://172.16.51.244:20000">172.16.51.244:20000</a>;transport=UDP><br>Content-Length: 0<br><br><br><br>_______________________________________________<br>--Bandwidth and Colocation provided by
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