[AsteriskBrasil] Problema com Nat SIP

Saulo Quinteiro sauloquinteiro em gmail.com
Sexta Maio 28 15:04:14 BRT 2010


Dai galera, to com problemas no nat com sip, tem nat nas duas pontas.
O rtp não passa.

Segue debug da ligação em uma fila o audio da fila não passa. (tirei o 
final dos ip para evitar engraçadinhos. querendo fazer brincadeiras)


<--- SIP read from 200.160.xxx.xxx:51151 --->
INVITE sip:7000 em 189.38.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 
172.30.1.64:51151;branch=z9hG4bK-d8754z-8d2c602a3a15574b-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:702 em 200.160.xxx.xxx:51151>
To: "7000"<sip:7000 em 189.38.xxx.xxx>
From: "702"<sip:702 em 189.38.xxx.xxx>;tag=027a0f27
Call-ID: ZWZkZGU2N2E2NzhlMTJjZjMwNDgzYjU0ZDliNjg4NDI.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, 
SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 261

v=0
o=- 2 2 IN IP4 172.30.1.64
s=CounterPath X-Lite 3.0
c=IN IP4 172.30.1.64
t=0 0
m=audio 13626 RTP/AVP 107 0 8 101
a=alt:1 1 : QdrDUv3A fbjZtXfs 172.30.1.64 13626
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
--- (12 headers 11 lines) ---
Sending to 200.160.xxx.xxx : 51151 (NAT)
Using INVITE request as basis request - 
ZWZkZGU2N2E2NzhlMTJjZjMwNDgzYjU0ZDliNjg4NDI.

<--- Reliably Transmitting (NAT) to 200.160.xxx.xxx:51151 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 
172.30.1.64:51151;branch=z9hG4bK-d8754z-8d2c602a3a15574b-1---d8754z-;received=200.160.xxx.xxx;rport=51151
From: "702"<sip:702 em 189.38.xxx.xxx>;tag=027a0f27
To: "7000"<sip:7000 em 189.38.xxx.xxx>;tag=as7f5973d2
Call-ID: ZWZkZGU2N2E2NzhlMTJjZjMwNDgzYjU0ZDliNjg4NDI.
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="17f78f20"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 
'ZWZkZGU2N2E2NzhlMTJjZjMwNDgzYjU0ZDliNjg4NDI.' in 32000 ms (Method: INVITE)
Found user '702'
vispbx01*CLI>
<--- SIP read from 200.160.xxx.xxx:51151 --->
ACK sip:7000 em 189.38.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 
172.30.1.64:51151;branch=z9hG4bK-d8754z-8d2c602a3a15574b-1---d8754z-;rport
To: "7000"<sip:7000 em 189.38.xxx.xxx>;tag=as7f5973d2
From: "702"<sip:702 em 189.38.xxx.xxx>;tag=027a0f27
Call-ID: ZWZkZGU2N2E2NzhlMTJjZjMwNDgzYjU0ZDliNjg4NDI.
CSeq: 1 ACK
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
vispbx01*CLI>
<--- SIP read from 200.160.xxx.xxx:51151 --->
INVITE sip:7000 em 189.38.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 
172.30.1.64:51151;branch=z9hG4bK-d8754z-0d75b23fb408b74f-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:702 em 200.160.xxx.xxx:51151>
To: "7000"<sip:7000 em 189.38.xxx.xxx>
From: "702"<sip:702 em 189.38.xxx.xxx>;tag=027a0f27
Call-ID: ZWZkZGU2N2E2NzhlMTJjZjMwNDgzYjU0ZDliNjg4NDI.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, 
SUBSCRIBE, INFO
Content-Type: application/sdp
Proxy-Authorization: Digest 
username="702",realm="asterisk",nonce="17f78f20",uri="sip:7000 em 189.38.xxx.xxx",response="f680a98eeae53cdb9e632d820f942017",algorithm=MD5
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 261

v=0
o=- 2 2 IN IP4 172.30.1.64
s=CounterPath X-Lite 3.0
c=IN IP4 172.30.1.64
t=0 0
m=audio 13626 RTP/AVP 107 0 8 101
a=alt:1 1 : QdrDUv3A fbjZtXfs 172.30.1.64 13626
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
--- (13 headers 11 lines) ---
Sending to 200.160.xxx.xxx : 51151 (NAT)
Using INVITE request as basis request - 
ZWZkZGU2N2E2NzhlMTJjZjMwNDgzYjU0ZDliNjg4NDI.
Found user '702'
Found RTP audio format 107
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found unknown media description format BV32 for ID 107
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc 
(ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 172.30.1.64:13626
Looking for 7000 in from-internal (domain 189.38.xxx.xxx)
list_route: hop: <sip:702 em 200.160.xxx.xxx:51151>

<--- Transmitting (NAT) to 200.160.xxx.xxx:51151 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 
172.30.1.64:51151;branch=z9hG4bK-d8754z-0d75b23fb408b74f-1---d8754z-;received=200.160.xxx.xxx;rport=51151
From: "702"<sip:702 em 189.38.xxx.xxx>;tag=027a0f27
To: "7000"<sip:7000 em 189.38.xxx.xxx>
Call-ID: ZWZkZGU2N2E2NzhlMTJjZjMwNDgzYjU0ZDliNjg4NDI.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:7000 em 192.168.0.102>
Content-Length: 0


<------------>
Audio is at 192.168.0.102 port 12906
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 200.160.xxx.xxx:51151 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
172.30.1.64:51151;branch=z9hG4bK-d8754z-0d75b23fb408b74f-1---d8754z-;received=200.160.xxx.xxx;rport=51151
From: "702"<sip:702 em 189.38.xxx.xxx>;tag=027a0f27
To: "7000"<sip:7000 em 189.38.xxx.xxx>;tag=as5616376f
Call-ID: ZWZkZGU2N2E2NzhlMTJjZjMwNDgzYjU0ZDliNjg4NDI.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:7000 em 192.168.0.102>
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 2752 2752 IN IP4 192.168.0.102
s=session
c=IN IP4 192.168.0.102
t=0 0
m=audio 12906 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
Really destroying SIP dialog 
'3b9e133d3529d1585aa5e1bd67f4dbb0 em 127.0.1.1' Method: INVITE
[May 28 13:56:30] WARNING[3058]: app_dial.c:1296 dial_exec_full: Unable 
to create channel of type 'SIP' (cause 20 - Unknown)
Really destroying SIP dialog 
'56bcb2c0785cc1e025834c291c9c6d04 em 127.0.1.1' Method: INVITE
[May 28 13:56:30] WARNING[3062]: app_dial.c:1296 dial_exec_full: Unable 
to create channel of type 'SIP' (cause 20 - Unknown)
Retransmitting #1 (NAT) to 200.160.xxx.xxx:51151:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
172.30.1.64:51151;branch=z9hG4bK-d8754z-0d75b23fb408b74f-1---d8754z-;received=200.160.xxx.xxx;rport=51151
From: "702"<sip:702 em 189.38.xxx.xxx>;tag=027a0f27
To: "7000"<sip:7000 em 189.38.xxx.xxx>;tag=as5616376f
Call-ID: ZWZkZGU2N2E2NzhlMTJjZjMwNDgzYjU0ZDliNjg4NDI.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:7000 em 192.168.0.102>
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 2752 2752 IN IP4 192.168.0.102
s=session
c=IN IP4 192.168.0.102
t=0 0
m=audio 12906 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
Retransmitting #2 (NAT) to 200.160.xxx.xxx:51151:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
172.30.1.64:51151;branch=z9hG4bK-d8754z-0d75b23fb408b74f-1---d8754z-;received=200.160.xxx.xxx;rport=51151
From: "702"<sip:702 em 189.38.xxx.xxx>;tag=027a0f27
To: "7000"<sip:7000 em 189.38.xxx.xxx>;tag=as5616376f
Call-ID: ZWZkZGU2N2E2NzhlMTJjZjMwNDgzYjU0ZDliNjg4NDI.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:7000 em 192.168.0.102>
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 2752 2752 IN IP4 192.168.0.102
s=session
c=IN IP4 192.168.0.102
t=0 0
m=audio 12906 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
Retransmitting #3 (NAT) to 200.160.xxx.xxx:51151:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
172.30.1.64:51151;branch=z9hG4bK-d8754z-0d75b23fb408b74f-1---d8754z-;received=200.160.xxx.xxx;rport=51151
From: "702"<sip:702 em 189.38.xxx.xxx>;tag=027a0f27
To: "7000"<sip:7000 em 189.38.xxx.xxx>;tag=as5616376f
Call-ID: ZWZkZGU2N2E2NzhlMTJjZjMwNDgzYjU0ZDliNjg4NDI.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:7000 em 192.168.0.102>
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 2752 2752 IN IP4 192.168.0.102
s=session
c=IN IP4 192.168.0.102
t=0 0
m=audio 12906 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
Really destroying SIP dialog 
'002ef8cd391fb9515bb446eb1f3bd90c em 127.0.1.1' Method: INVITE
[May 28 13:56:35] WARNING[3074]: app_dial.c:1296 dial_exec_full: Unable 
to create channel of type 'SIP' (cause 20 - Unknown)
Really destroying SIP dialog 
'7fd5945017d1ecc7701cbcd30dc2e9f8 em 127.0.1.1' Method: INVITE
[May 28 13:56:35] WARNING[3070]: app_dial.c:1296 dial_exec_full: Unable 
to create channel of type 'SIP' (cause 20 - Unknown)
vispbx01*CLI>
<--- SIP read from 200.160.xxx.xxx:51151 --->



<------------->


Ja configurei externip=189.38.xxx.xxx , 
localhost=192.168.0.0/255.255.255.0, nat=yes, canreinvite=no
Agradeço qualquer ajuda.

Att,

-- 
Saulo Quinteiro dos Santos
Bacharel em Ciências da Computação UFPR
Cel: 	(041) 9927-5236
Com:	(041) 2141-9567



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