Requesting assistance troubleshooting one way voice


#1

Hello…

I am trying to setup communication between two extensions located
at physically different locations. The Asterisk server and extension P
are at the same LAN/location while extension Q is at the remote site.
Both sites use NAT. However, at both sites, we are mapping a static
public IP to the internal private IP. Here’s the simplified layout:


           P--+
              |    67.41.155.244       
              +--[router_a]----//----[router_b]---Q
              |               61.247.239.228 
           *--+

  P        = 172.27.72.99
  *        = 172.27.72.53
  router_a = 172.27.72.254 (nat 67.41.155.244)
  Q        = 192.168.1.10
  router_b = 192.168.1.1 (nat 61.247.239.228)

When I call Q from P (via *), I can hear the person at remote end but
then cannot hear me. Likewise, when Q calls P, they can hear me, but
I cannot hear them. I am suspecting a problem with RTP, but so far I have not been able to pinpoint the issue. Could someone review the following debug data and see if I’m missing something obvious? This was captured when P calls Q. Thank you very much for your time and assistance.

Regards

serenity*CLI> sip debug   
SIP Debugging enabled
serenity*CLI> 
<-- SIP read from 172.27.72.99:5060: 
INVITE sip:7601@172.27.72.53:5060 SIP/2.0
Via: SIP/2.0/UDP 172.27.72.99:5060;branch=z9hG4bKc5bf655bc
Max-Forwards: 70
Content-Length: 253
To: sip:7601@172.27.72.53:5060
From: Rajib Rashid <sip:9999@172.27.72.53:5060>;tag=65c268c6e797191
Call-ID: 4fa9ddeec541fe113d7a1d4757a0df6a@172.27.72.53
CSeq: 883949197 INVITE
Route: <sip:7601@172.27.72.53:5060;lr>
Supported: timer
Min-SE: 5
Allow-Events: talk, hold
Allow: INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,MESSAGE
Content-Type: application/sdp
Contact: Rajib Rashid <sip:9999@172.27.72.99:5060;transport=udp>
Supported: replaces
User-Agent: optiPoint 400 standard

v=0
o=MxSIP 0 1421231343 IN IP4 172.27.72.99
s=SIP Call
c=IN IP4 172.27.72.99
t=0 0
m=audio 5004 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

--- (17 headers 12 lines)---
Using INVITE request as basis request - 4fa9ddeec541fe113d7a1d4757a0df6a@172.27.72.53
Sending to 172.27.72.99 : 5060 (NAT)
Reliably Transmitting (no NAT) to 172.27.72.99:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 172.27.72.99:5060;branch=z9hG4bKc5bf655bc
From: Rajib Rashid <sip:9999@172.27.72.53:5060>;tag=65c268c6e797191
To: sip:7601@172.27.72.53:5060;tag=as4075ab71
Call-ID: 4fa9ddeec541fe113d7a1d4757a0df6a@172.27.72.53
CSeq: 883949197 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:7601@172.27.72.53>
Proxy-Authenticate: Digest realm="asterisk", nonce="34e178ec" 
Content-Length: 0


---
Scheduling destruction of call '4fa9ddeec541fe113d7a1d4757a0df6a@172.27.72.53' in 15000 ms
Found user '9999'
serenity*CLI> 
<-- SIP read from 172.27.72.99:5060: 
ACK sip:7601@172.27.72.53:5060 SIP/2.0
Via: SIP/2.0/UDP 172.27.72.99:5060;branch=z9hG4bKc5bf655bc
Max-Forwards: 70
Content-Length: 0
To: sip:7601@172.27.72.53:5060;tag=as4075ab71
From: Rajib Rashid <sip:9999@172.27.72.53:5060>;tag=65c268c6e797191
Call-ID: 4fa9ddeec541fe113d7a1d4757a0df6a@172.27.72.53
CSeq: 883949197 ACK
Route: <sip:7601@172.27.72.53:5060;lr>
User-Agent: optiPoint 400 standard


--- (10 headers 0 lines)---
serenity*CLI> 
<-- SIP read from 172.27.72.99:5060: 
INVITE sip:7601@172.27.72.53:5060 SIP/2.0
Via: SIP/2.0/UDP 172.27.72.99:5060;branch=z9hG4bKa3277e577
Max-Forwards: 70
Content-Length: 253
To: sip:7601@172.27.72.53:5060
From: Rajib Rashid <sip:9999@172.27.72.53:5060>;tag=65c268c6e797191
Call-ID: 4fa9ddeec541fe113d7a1d4757a0df6a@172.27.72.53
CSeq: 883949198 INVITE
Route: <sip:7601@172.27.72.53:5060;lr>
Supported: timer
Min-SE: 5
Allow-Events: talk, hold
Allow: INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,MESSAGE
Contact: Rajib Rashid <sip:9999@172.27.72.99:5060;transport=udp>
Content-Type: application/sdp
Proxy-Authorization:Digest response="2a716da9f3b3ebaa07762cdc94f481d3",username="9999",realm="asterisk",nonce="34e178ec",uri="sip:7601@172.27.72.53:5060"
Supported: replaces
User-Agent: optiPoint 400 standard

v=0
o=MxSIP 0 1421231343 IN IP4 172.27.72.99
s=SIP Call
c=IN IP4 172.27.72.99
t=0 0
m=audio 5004 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

--- (18 headers 12 lines)---
Using INVITE request as basis request - 4fa9ddeec541fe113d7a1d4757a0df6a@172.27.72.53
Sending to 172.27.72.99 : 5060 (non-NAT)
Found user '9999'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 172.27.72.99:5004
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format telephone-event
Capabilities: us - 0x8 (alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 7601 in home
list_route: hop: <sip:9999@172.27.72.99:5060;transport=udp>
Transmitting (no NAT) to 172.27.72.99:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.27.72.99:5060;branch=z9hG4bKa3277e577
From: Rajib Rashid <sip:9999@172.27.72.53:5060>;tag=65c268c6e797191
To: sip:7601@172.27.72.53:5060
Call-ID: 4fa9ddeec541fe113d7a1d4757a0df6a@172.27.72.53
CSeq: 883949198 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:7601@172.27.72.53>
Content-Length: 0


---
We're at 67.41.155.244 port 5008
Answering/Requesting with root capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 10 lines
Reliably Transmitting (NAT) to 61.247.239.228:5060:
INVITE sip:7601@61.247.239.228 SIP/2.0
Via: SIP/2.0/UDP 67.41.155.244:5060;branch=z9hG4bK36c1a4ed;rport
From: "Rajib Rashid" <sip:9999@67.41.155.244>;tag=as16a92105
To: <sip:7601@61.247.239.228>
Contact: <sip:9999@67.41.155.244>
Call-ID: 3bcf79273ecf34e51892036857de43e3@67.41.155.244
CSeq: 102 INVITE
User-Agent: Asterisk
Date: Tue, 13 Dec 2005 05:50:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 215

v=0
o=root 6084 6084 IN IP4 67.41.155.244
s=session
c=IN IP4 67.41.155.244
t=0 0
m=audio 5008 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
serenity*CLI> 
<-- SIP read from 61.247.239.228:5060: 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 67.41.155.244:5060;branch=z9hG4bK36c1a4ed;rport=5060;received=67.41.155.244
From: "Rajib Rashid" <sip:9999@67.41.155.244>;tag=as16a92105
To: <sip:7601@61.247.239.228>
Call-ID: 3bcf79273ecf34e51892036857de43e3@67.41.155.244
CSeq: 102 INVITE
Contact: <sip:7601@61.247.239.228:5060>
Content-Length: 0


--- (8 headers 0 lines)---
serenity*CLI> 
<-- SIP read from 61.247.239.228:5060: 
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 67.41.155.244:5060;branch=z9hG4bK36c1a4ed;rport=5060;received=67.41.155.244
From: "Rajib Rashid" <sip:9999@67.41.155.244>;tag=as16a92105
To: <sip:7601@61.247.239.228>;tag=-956557715
Call-ID: 3bcf79273ecf34e51892036857de43e3@67.41.155.244
CSeq: 102 INVITE
Contact:  <sip:7601@61.247.239.228:5060>
Content-Length: 0


--- (8 headers 0 lines)---
Transmitting (no NAT) to 172.27.72.99:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.27.72.99:5060;branch=z9hG4bKa3277e577
From: Rajib Rashid <sip:9999@172.27.72.53:5060>;tag=65c268c6e797191
To: sip:7601@172.27.72.53:5060;tag=as5a898bfa
Call-ID: 4fa9ddeec541fe113d7a1d4757a0df6a@172.27.72.53
CSeq: 883949198 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:7601@172.27.72.53>
Content-Length: 0


---
serenity*CLI> 
<-- SIP read from 61.247.239.228:5060: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.41.155.244:5060;branch=z9hG4bK36c1a4ed;rport=5060;received=67.41.155.244
From: "Rajib Rashid" <sip:9999@67.41.155.244>;tag=as16a92105
To: <sip:7601@61.247.239.228>;tag=-956557715
Call-ID: 3bcf79273ecf34e51892036857de43e3@67.41.155.244
CSeq: 102 INVITE
Contact:  <sip:7601@61.247.239.228:5060>
user-agent: NGAVFXS/1.08.DK.01.08Sep 30 2005
Allow: INVITE, ACK, BYE, CANCEL, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 195

v=0
o=7601 0 0 IN IP4 61.247.239.228
s=SIP Call
c=IN IP4 61.247.239.228
t=0 0
m=audio 11631 RTP/AVP 8 101
a=ptime:80
a=rtpmap:8 PCMA/8000/1
a=fmtp:101 0-16
a=rtpmap:101 telephone-event

--- (11 headers 10 lines)---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 61.247.239.228:11631
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:7601@61.247.239.228:5060>
set_destination: Parsing <sip:7601@61.247.239.228:5060> for address/port to send to
set_destination: set destination to 61.247.239.228, port 5060
Transmitting (NAT) to 61.247.239.228:5060:
ACK sip:7601@61.247.239.228:5060 SIP/2.0
Via: SIP/2.0/UDP 67.41.155.244:5060;branch=z9hG4bK28972209;rport
From: "Rajib Rashid" <sip:9999@67.41.155.244>;tag=as16a92105
To: <sip:7601@61.247.239.228>;tag=-956557715
Contact: <sip:9999@67.41.155.244>
Call-ID: 3bcf79273ecf34e51892036857de43e3@67.41.155.244
CSeq: 102 ACK
User-Agent: Asterisk
Content-Length: 0


---
We're at 172.27.72.53 port 5006
Answering with preferred capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
Reliably Transmitting (no NAT) to 172.27.72.99:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.27.72.99:5060;branch=z9hG4bKa3277e577
From: Rajib Rashid <sip:9999@172.27.72.53:5060>;tag=65c268c6e797191
To: sip:7601@172.27.72.53:5060;tag=as5a898bfa
Call-ID: 4fa9ddeec541fe113d7a1d4757a0df6a@172.27.72.53
CSeq: 883949198 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:7601@172.27.72.53>
Content-Type: application/sdp
Content-Length: 213

v=0
o=root 6084 6084 IN IP4 172.27.72.53
s=session
c=IN IP4 172.27.72.53
t=0 0
m=audio 5006 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
serenity*CLI> 
<-- SIP read from 172.27.72.99:5060: 
ACK sip:7601@172.27.72.53 SIP/2.0
Via: SIP/2.0/UDP 172.27.72.99:5060;branch=z9hG4bK65cc9b474
Max-Forwards: 70
Content-Length: 0
To: sip:7601@172.27.72.53:5060;tag=as5a898bfa
From: Rajib Rashid <sip:9999@172.27.72.53:5060>;tag=65c268c6e797191
Call-ID: 4fa9ddeec541fe113d7a1d4757a0df6a@172.27.72.53
CSeq: 883949198 ACK
Contact: Rajib Rashid <sip:9999@172.27.72.99:5060;transport=udp>
Proxy-Authorization:Digest response="2a716da9f3b3ebaa07762cdc94f481d3",username="9999",realm="asterisk",nonce="34e178ec",uri="sip:7601@172.27.72.53:5060"
User-Agent: optiPoint 400 standard


--- (11 headers 0 lines)---
11 headers, 0 lines
Reliably Transmitting (no NAT) to 61.247.239.228:5060:
OPTIONS sip:7980@61.247.239.228:5060 SIP/2.0
Via: SIP/2.0/UDP 67.41.155.244:5060;branch=z9hG4bK0ba6eaa3
From: "asterisk" <sip:asterisk@67.41.155.244>;tag=as360cfd3c
To: <sip:7980@61.247.239.228:5060>
Contact: <sip:asterisk@67.41.155.244>
Call-ID: 4a603369166ccc0248ef63ff786d5921@67.41.155.244
CSeq: 102 OPTIONS
User-Agent: Asterisk
Date: Tue, 13 Dec 2005 05:51:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Length: 0


---
serenity*CLI> 
<-- SIP read from 61.247.239.228:5060: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.41.155.244:5060;branch=z9hG4bK0ba6eaa3
From: "asterisk" <sip:asterisk@67.41.155.244>;tag=as360cfd3c
To: <sip:7980@61.247.239.228:5060>
Call-ID: 4a603369166ccc0248ef63ff786d5921@67.41.155.244
CSeq: 102 OPTIONS
Contact: <sip:7980@192.168.1.10:5060>
Allow: INVITE, ACK, BYE, CANCEL, REFER, NOTIFY
Accept: text/plian, application/sdp
Accept-Encoding: identity
Content-Length: 358

v=0
o=7980 0 0 IN IP4 192.168.1.10
s=SIP Call
c=IN IP4 
t=0 0
m=audio 5004 RTP/AVP 0 1 22 5 12 13 14 4 9 10 11 3 7 8 2 6 31 30 29 17 17 17 17 28 27 16 15
a=rtpmap:0 PCMA/8000/1
a=rtpmap:1 PCMU/8000/1
a=rtpmap:17 G729/8000/1
a=rtpmap:17 G729/8000/1
a=rtpmap:17 G729/8000/1
a=rtpmap:17 G729/8000/1
a=rtpmap:16 G723/8000/1
a=rtpmap:15 G723/8000/1

--- (11 headers 14 lines)---
Destroying call '4a603369166ccc0248ef63ff786d5921@67.41.155.244'
serenity*CLI> 
<-- SIP read from 172.27.72.99:5060: 
BYE sip:7601@172.27.72.53 SIP/2.0
Via: SIP/2.0/UDP 172.27.72.99:5060;branch=z9hG4bK07ed512f7
Max-Forwards: 70
Content-Length: 0
To: sip:7601@172.27.72.53:5060;tag=as5a898bfa
From: Rajib Rashid <sip:9999@172.27.72.53:5060>;tag=65c268c6e797191
Call-ID: 4fa9ddeec541fe113d7a1d4757a0df6a@172.27.72.53
CSeq: 883949199 BYE
Supported: timer
Proxy-Authorization:Digest response="fc0a226f39e6dee99f2d29317f9183ca",username="9999",realm="asterisk",nonce="34e178ec",uri="sip:7601@172.27.72.53"
Supported: replaces
User-Agent: optiPoint 400 standard


--- (12 headers 0 lines)---
Sending to 172.27.72.99 : 5060 (non-NAT)
Transmitting (no NAT) to 172.27.72.99:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.27.72.99:5060;branch=z9hG4bK07ed512f7
From: Rajib Rashid <sip:9999@172.27.72.53:5060>;tag=65c268c6e797191
To: sip:7601@172.27.72.53:5060;tag=as5a898bfa
Call-ID: 4fa9ddeec541fe113d7a1d4757a0df6a@172.27.72.53
CSeq: 883949199 BYE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:7601@172.27.72.53>
Content-Length: 0
X-Asterisk-HangupCause:: Normal Clearing


---
set_destination: Parsing <sip:7601@61.247.239.228:5060> for address/port to send to
set_destination: set destination to 61.247.239.228, port 5060
Reliably Transmitting (NAT) to 61.247.239.228:5060:
BYE sip:7601@61.247.239.228:5060 SIP/2.0
Via: SIP/2.0/UDP 67.41.155.244:5060;branch=z9hG4bK735ace14;rport
From: "Rajib Rashid" <sip:9999@67.41.155.244>;tag=as16a92105
To: <sip:7601@61.247.239.228>;tag=-956557715
Contact: <sip:9999@67.41.155.244>
Call-ID: 3bcf79273ecf34e51892036857de43e3@67.41.155.244
CSeq: 103 BYE
User-Agent: Asterisk
Content-Length: 0


---
Destroying call '4fa9ddeec541fe113d7a1d4757a0df6a@172.27.72.53'
serenity*CLI> 
<-- SIP read from 61.247.239.228:5060: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.41.155.244:5060;branch=z9hG4bK735ace14;rport=5060;received=67.41.155.244
From: "Rajib Rashid" <sip:9999@67.41.155.244>;tag=as16a92105
To: <sip:7601@61.247.239.228>;tag=-956557715
Call-ID: 3bcf79273ecf34e51892036857de43e3@67.41.155.244
CSeq: 103 BYE
Contact:  <sip:7601@61.247.239.228:5060>
user-agent: NGAVFXS/1.08.DK.01.08Sep 30 2005
Content-Length: 0


--- (9 headers 0 lines)---
Destroying call '3bcf79273ecf34e51892036857de43e3@67.41.155.244'

#2

are you port forwarding the full RTP port range?


#3

Yes… the internal IP address of the * server has been mapped to a public IP and same with the and remote client.


#4

Have you got “nat” and “externip” configured properly in sip.conf? (It probably wouldn’t work at all if you hadn’t, but i’m just checking.)


#5

sip.conf has:

Please let me know if anything jumps out as being odd. Thanks!