[asterisk-users] Attempting native bridge of

Victor Toofic toofics at gmail.com
Wed Nov 15 16:10:10 MST 2006 

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

-------------- next part --------------
----------------------------------------------- 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

-------------- next part --------------
----------------------------------------------- 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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