[Asterisk-Users] Sip and One Way Audio

Tue Aug 12 07:39:16 MST 2003

Were having the same problem, but the other way around.

We have some SIP UA's behind a NAT firewall.
This problem only applies to a particular brand of TA's, the others 
(Cisco ATA's) works fine.

1) Calls from the outside to the inside
    Audio works fine in both directions

2) Calls from the inside to the outside
    Audio works from the PSTN (PRI) to the SIP UA, but not the other way 
around. ("The-other-way-around-problem")
    The call gets dropped in about five seconds and then * (or the UA, 
not sure) tries to reinitiate the connection, and the call is set up again.

The SIP debug trace follows.

/Fredrik

Retransmitting #3 (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
217.210.59.204:6010;branch=z9hG4bK1031098166770781;received=217.210.59.204:6010
From: sip:u0004 at 217.210.59.204;tag=1031098166762209
To: sip:020550010 at 62.209.162.162;tag=as62183eb3
Call-ID: 1031098166764920 at 217.210.59.204
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Contact: <sip:020550010 at 62.209.162.162>
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 17159 17159 IN IP4 62.209.162.162
s=session
c=IN IP4 62.209.162.162
t=0 0
m=audio 15550 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
@
a=f
 to 217.210.59.204:6010
Sip read:
INVITE sip:020550010 at 62.209.162.162 SIP/2.0
Via: SIP/2.0/UDP 217.210.59.204:6010;branch=z9hG4bK1031098166770781
Max-Forwards: 70
To: sip:020550010 at 62.209.162.162
From: sip:u0004 at 217.210.59.204;tag=1031098166762209
Call-ID: 1031098166764920 at 217.210.59.204
CSeq: 2 INVITE
Contact: sip:u0004 at 217.210.59.204:6010
User-Agent: i3micro Vood (122s_1_3_1_5_UVM_swe 122s_1_3_1_5, R2D)
Content-Length: 209
Content-Type: application/sdp

v=0
o=- 1031098166 1031098166 IN IP4 217.210.59.204
s=-
c=IN IP4 217.210.59.204
t=0 0
m=audio 10002 RTP/AVP 8 0 3
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=ptime:30
a=sendrecv

11 headers, 11 lines
Ignoring this request
We're at 62.209.162.162 port 15550
Answering with preferred capability 4
Answering with preferred capability 8
Answering with capability 2
Answering with non-codec capability 1
Reliably Transmitting (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
217.210.59.204:6010;branch=z9hG4bK1031098166770781;received=217.210.59.204:6010
From: sip:u0004 at 217.210.59.204;tag=1031098166762209
To: sip:020550010 at 62.209.162.162;tag=as62183eb3
Call-ID: 1031098166764920 at 217.210.59.204
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Contact: <sip:020550010 at 62.209.162.162>
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 14852 14852 IN IP4 62.209.162.162
s=session
c=IN IP4 62.209.162.162
t=0 0
m=audio 15550 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

 to 217.210.59.204:6010
Retransmitting #3 (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
217.210.59.204:6010;branch=z9hG4bK1031098166770781;received=217.210.59.204:6010
From: sip:u0004 at 217.210.59.204;tag=1031098166762209
To: sip:020550010 at 62.209.162.162;tag=as62183eb3
Call-ID: 1031098166764920 at 217.210.59.204
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Contact: <sip:020550010 at 62.209.162.162>
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 14852 14852 IN IP4 62.209.162.162
s=session
c=IN IP4 62.209.162.162
t=0 0
m=audio 15550 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

 to 217.210.59.204:6010
Retransmitting #4 (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
217.210.59.204:6010;branch=z9hG4bK1031098166770781;received=217.210.59.204:6010
From: sip:u0004 at 217.210.59.204;tag=1031098166762209
To: sip:020550010 at 62.209.162.162;tag=as62183eb3
Call-ID: 1031098166764920 at 217.210.59.204
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Contact: <sip:020550010 at 62.209.162.162>
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 17159 17159 IN IP4 62.209.162.162
s=session
c=IN IP4 62.209.162.162
t=0 0
m=audio 15550 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
@
a=f
 to 217.210.59.204:6010
Retransmitting #1 (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
217.210.59.204:6010;branch=z9hG4bK1031098166770781;received=217.210.59.204:6010
From: sip:u0004 at 217.210.59.204;tag=1031098166762209
To: sip:020550010 at 62.209.162.162;tag=as62183eb3
Call-ID: 1031098166764920 at 217.210.59.204
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Contact: <sip:020550010 at 62.209.162.162>
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 14852 14852 IN IP4 62.209.162.162
s=session
c=IN IP4 62.209.162.162
t=0 0
m=audio 15550 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

 to 217.210.59.204:6010
Retransmitting #4 (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
217.210.59.204:6010;branch=z9hG4bK1031098166770781;received=217.210.59.204:6010
From: sip:u0004 at 217.210.59.204;tag=1031098166762209
To: sip:020550010 at 62.209.162.162;tag=as62183eb3
Call-ID: 1031098166764920 at 217.210.59.204
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Contact: <sip:020550010 at 62.209.162.162>
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 14852 14852 IN IP4 62.209.162.162
s=session
c=IN IP4 62.209.162.162
t=0 0
m=audio 15550 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

 to 217.210.59.204:6010
Retransmitting #5 (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
217.210.59.204:6010;branch=z9hG4bK1031098166770781;received=217.210.59.204:6010
From: sip:u0004 at 217.210.59.204;tag=1031098166762209
To: sip:020550010 at 62.209.162.162;tag=as62183eb3
Call-ID: 1031098166764920 at 217.210.59.204
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Contact: <sip:020550010 at 62.209.162.162>
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 17159 17159 IN IP4 62.209.162.162
s=session
c=IN IP4 62.209.162.162
t=0 0
m=audio 15550 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
@
a=f
 to 217.210.59.204:6010
Retransmitting #2 (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
217.210.59.204:6010;branch=z9hG4bK1031098166770781;received=217.210.59.204:6010
From: sip:u0004 at 217.210.59.204;tag=1031098166762209
To: sip:020550010 at 62.209.162.162;tag=as62183eb3
Call-ID: 1031098166764920 at 217.210.59.204
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Contact: <sip:020550010 at 62.209.162.162>
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 14852 14852 IN IP4 62.209.162.162
s=session
c=IN IP4 62.209.162.162
t=0 0
m=audio 15550 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

 to 217.210.59.204:6010
Retransmitting #5 (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
217.210.59.204:6010;branch=z9hG4bK1031098166770781;received=217.210.59.204:6010
From: sip:u0004 at 217.210.59.204;tag=1031098166762209
To: sip:020550010 at 62.209.162.162;tag=as62183eb3
Call-ID: 1031098166764920 at 217.210.59.204
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Contact: <sip:020550010 at 62.209.162.162>
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 14852 14852 IN IP4 62.209.162.162
s=session
c=IN IP4 62.209.162.162
t=0 0
m=audio 15550 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

 to 217.210.59.204:6010
WARNING[5126]: File chan_sip.c, Line 415 (retrans_pkt): Maximum retries 
exceeded on call 1031098166764920 at 217.210.59.204 for seqno 2 (Response)
    -- Hungup 'Zap/1-1'
  == Spawn extension (telavox, 020550010, 1) exited non-zero on 
'SIP/u0004-b3c1'
set_destination: Parsing <sip:u0004 at 217.210.59.204:6010> for 
address/port to send to
set_destination: set destination to 217.210.59.204, port 6010
Reliably Transmitting:
BYE sip:u0004 at 217.210.59.204 SIP/2.0
Via: SIP/2.0/UDP 62.209.162.162:5060;branch=z9hG4bK7d974523
From: sip:020550010 at 62.209.162.162;tag=as62183eb3
To: sip:u0004 at 217.210.59.204;tag=1031098166762209
Contact: <sip:020550010 at 62.209.162.162>
Call-ID: 1031098166764920 at 217.210.59.204
CSeq: 102 BYE
User-Agent: Asterisk PBX
Content-Length: 0

 (NAT) to 217.210.59.204:6010
Retransmitting #3 (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
217.210.59.204:6010;branch=z9hG4bK1031098166770781;received=217.210.59.204:6010
From: sip:u0004 at 217.210.59.204;tag=1031098166762209
To: sip:020550010 at 62.209.162.162;tag=as62183eb3
Call-ID: 1031098166764920 at 217.210.59.204
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Contact: <sip:020550010 at 62.209.162.162>
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 14852 14852 IN IP4 62.209.162.162
s=session
c=IN IP4 62.209.162.162
t=0 0
m=audio 15550 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

 to 217.210.59.204:6010
WARNING[5126]: File chan_sip.c, Line 415 (retrans_pkt): Maximum retries 
exceeded on call 1031098166764920 at 217.210.59.204 for seqno 2 (Response)
Sip read:
INVITE sip:020550010 at 62.209.162.162 SIP/2.0
Via: SIP/2.0/UDP 217.210.59.204:6010;branch=z9hG4bK1031098166770781
Max-Forwards: 70
To: sip:020550010 at 62.209.162.162
From: sip:u0004 at 217.210.59.204;tag=1031098166762209
Call-ID: 1031098166764920 at 217.210.59.204
CSeq: 2 INVITE
Contact: sip:u0004 at 217.210.59.204:6010
User-Agent: i3micro Vood (122s_1_3_1_5_UVM_swe 122s_1_3_1_5, R2D)
Content-Length: 209
Content-Type: application/sdp

v=0
o=- 1031098166 1031098166 IN IP4 217.210.59.204
s=-
c=IN IP4 217.210.59.204
t=0 0
m=audio 10002 RTP/AVP 8 0 3
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=ptime:30
a=sendrecv

11 headers, 11 lines
Using latest request as basis request
Sending to 217.210.59.204 : 6010 (non-NAT)
Found audio format 8
Found audio format 0
Found audio format 3
Found description format PCMA
Found description format PCMU
Found description format GSM
Capabilities: us - 524558, them - 14/0, combined - 14
Non-codec capabilities: us - 1, them - 0, combined - 0
Looking for 020550010 in telavox
list_route: hop: <sip:u0004 at 217.210.59.204:6010>
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 
217.210.59.204:6010;branch=z9hG4bK1031098166770781;received=217.210.59.204:6010
From: sip:u0004 at 217.210.59.204;tag=1031098166762209
To: sip:020550010 at 62.209.162.162;tag=as18126b1c
Call-ID: 1031098166764920 at 217.210.59.204
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Contact: <sip:020550010 at 62.209.162.162>
Content-Length: 0

 to 217.210.59.204:6010
    -- Executing AGI("SIP/u0004-3201", "std-outgoing.php") in new stack
    -- Launched AGI Script 
/usr/local/asterisk/var/lib/asterisk/agi-bin/std-outgoing.php
    -- AGI Script Executing Application: (Dial) Options: 
(Zap/g2/020550010|180|Tr)
    -- Called g2/020550010
Transmitting (NAT):
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 
217.210.59.204:6010;branch=z9hG4bK1031098166770781;received=217.210.59.204:6010
From: sip:u0004 at 217.210.59.204;tag=1031098166762209
To: sip:020550010 at 62.209.162.162;tag=as18126b1c
Call-ID: 1031098166764920 at 217.210.59.204
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Contact: <sip:020550010 at 62.209.162.162>
Content-Length: 0

 to 217.210.59.204:6010
    -- Zap/1-1 is ringing
We're at 62.209.162.162 port 10210
Answering with preferred capability 4
Answering with preferred capability 8
Answering with capability 2
Answering with non-codec capability 1
Transmitting (NAT):
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 
217.210.59.204:6010;branch=z9hG4bK1031098166770781;received=217.210.59.204:6010
From: sip:u0004 at 217.210.59.204;tag=1031098166762209
To: sip:020550010 at 62.209.162.162;tag=as18126b1c
Call-ID: 1031098166764920 at 217.210.59.204
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Contact: <sip:020550010 at 62.209.162.162>
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 17163 17163 IN IP4 62.209.162.162
s=session
c=IN IP4 62.209.162.162
t=0 0
m=audio 10210 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

 to 217.210.59.204:6010
    -- Zap/1-1 answered SIP/u0004-3201
We're at 62.209.162.162 port 10210
Answering with preferred capability 4
Answering with preferred capability 8
Answering with capability 2
Answering with non-codec capability 1
Reliably Transmitting (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
217.210.59.204:6010;branch=z9hG4bK1031098166770781;received=217.210.59.204:6010
From: sip:u0004 at 217.210.59.204;tag=1031098166762209
To: sip:020550010 at 62.209.162.162;tag=as18126b1c
Call-ID: 1031098166764920 at 217.210.59.204
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Contact: <sip:020550010 at 62.209.162.162>
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 17163 17163 IN IP4 62.209.162.162
s=session
c=IN IP4 62.209.162.162
t=0 0
m=audio 10210 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

 to 217.210.59.204:6010
Retransmitting #1 (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
217.210.59.204:6010;branch=z9hG4bK1031098166770781;received=217.210.59.204:6010
From: sip:u0004 at 217.210.59.204;tag=1031098166762209
To: sip:020550010 at 62.209.162.162;tag=as18126b1c
Call-ID: 1031098166764920 at 217.210.59.204
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Contact: <sip:020550010 at 62.209.162.162>
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 17163 17163 IN IP4 62.209.162.162
s=session
c=IN IP4 62.209.162.162
t=0 0
m=audio 10210 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

 to 217.210.59.204:6010
Sip read:
SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/UDP 62.209.162.162:5060;branch=z9hG4bK7d974523
To: sip:u0004 at 217.210.59.204;tag=1031098166762209
From: sip:020550010 at 62.209.162.162;tag=as62183eb3
Call-ID: 1031098166764920 at 217.210.59.204
CSeq: 102 BYE
User-Agent: i3micro Vood (122s_1_3_1_5_UVM_swe 122s_1_3_1_5, R2D)
Content-Length: 0

8 headers, 0 lines
Retransmitting #2 (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
217.210.59.204:6010;branch=z9hG4bK1031098166770781;received=217.210.59.204:6010
From: sip:u0004 at 217.210.59.204;tag=1031098166762209
To: sip:020550010 at 62.209.162.162;tag=as18126b1c
Call-ID: 1031098166764920 at 217.210.59.204
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Contact: <sip:020550010 at 62.209.162.162>
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 17163 17163 IN IP4 62.209.162.162
s=session
c=IN IP4 62.209.162.162
t=0 0
m=audio 10210 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

 to 217.210.59.204:6010
Retransmitting #3 (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
217.210.59.204:6010;branch=z9hG4bK1031098166770781;received=217.210.59.204:6010
From: sip:u0004 at 217.210.59.204;tag=1031098166762209
To: sip:020550010 at 62.209.162.162;tag=as18126b1c
Call-ID: 1031098166764920 at 217.210.59.204
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Contact: <sip:020550010 at 62.209.162.162>
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 17163 17163 IN IP4 62.209.162.162
s=session
c=IN IP4 62.209.162.162
t=0 0
m=audio 10210 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

Nathan Littlepage wrote:

>I believe that is NAT. Like many firewalls traffic going from behind the
>firewall is allowed. Traffic returning to the firewall may be blocked if
>the port is not open. Thus one way audio.
>
>  
>
>>-----Original Message-----
>>From: asterisk-users-admin at lists.digium.com 
>>[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Steve Lane
>>Sent: Tuesday, August 12, 2003 8:30 AM
>>To: asterisk-users at lists.digium.com
>>Subject: RE: [Asterisk-Users] Sip and One Way Audio
>>
>>
>>Would the firewall pose a problem? I thought Asterisk had the solution
>>for working behind a firewall? 
>>
>>-----Original Message-----
>>From: asterisk-users-admin at lists.digium.com
>>[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Nathan
>>Littlepage
>>Sent: Tuesday, August 12, 2003 8:19 AM
>>To: asterisk-users at lists.digium.com
>>Subject: RE: [Asterisk-Users] Sip and One Way Audio
>>
>>Smells like it.
>>
>>    
>>
>>>-----Original Message-----
>>>From: asterisk-users-admin at lists.digium.com 
>>>[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of 
>>>Dave Cotton
>>>Sent: Tuesday, August 12, 2003 8:14 AM
>>>To: Asterisk List
>>>Subject: Re: [Asterisk-Users] Sip and One Way Audio
>>>
>>>
>>>On Tue, 2003-08-12 at 15:05, Adelino Baena wrote:
>>>      
>>>
>>>>Dears
>>>> 
>>>>I got one way audio using SIP, X-Ten and SIP -> POTS 
>>>>        
>>>>
>>>integration. Does
>>>      
>>>
>>>>anybody knows some bug ?
>>>>        
>>>>
>>>Do I smell a firewall?
>>>
>>>-- 
>>>Dave Cotton <dcotton at linuxautrement.com>
>>>
>>>_______________________________________________
>>>Asterisk-Users mailing list
>>>Asterisk-Users at lists.digium.com
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