[asterisk-users] Attempting native bridge of

Thu Nov 16 05:58:19 MST 2006

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 .

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