RTP stream does not open between peers

Good day,

I am happy to join this great community and hope that this mature community will help me learn better.

I have two peers with PJSIP accounts connected to the Raspberry Pi. I can call from one account to the other. However, no communiction/voice transmission occurs between the two. The ASterisk CLI reads as follows:

– PJSIP/24-0000000f Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
– Called PJSIP/24/sip:24@192.168.200.210:5060;ob 
== Using SIP RTP Audio TOS bits 184 
== Using SIP RTP Audio TOS bits 184 in TCLASS field. 
== Using SIP RTP Audio CoS mark 5 
– PJSIP/24-0000000f answered PJSIP/23-0000000e 
> 0x7410d310 – Strict RTP learning after remote address set to: 192.168.200.210:4006
> 0x74113cd0 – Strict RTP learning after remote address set to: 192.168.200.230:4006
– Channel PJSIP/24-0000000f joined 'simple_bridge' basic-bridge <0e0ed803-5d80-4594-b8a0-4d330d89d0f6>
– Channel PJSIP/23-0000000e joined 'simple_bridge' basic-bridge <0e0ed803-5d80-4594-b8a0-4d330d89d0f6> 
> 0x7410d310 – Strict RTP switching to RTP target address 192.168.200.210:4006 as source
> 0x7410d310 – Strict RTP learning complete - Locking on source address 192.168.200.210:4006

But no communication happens. The problem is that I don’t see that any stream has been opened or selected. Like:

– Strict RTP qualifying stream type: audio

I see this informartion when I call a peer from another PJSIP account and voice transmission occurs.

Am I missing something?

Thank you in advance for any help.

This looks like it could be FreePBX, which has its own forum.

To understand what is going on one would probably need to see the SDP from the “pjsip set logger on” output, in the full log, and also the result of “rtp set debug on”.

Audio problems usually relate to firewalls or the use of NAT.

Please mark up logs posted in-inline as pre-formatted text, so that the forum doesn’t reflow them, or treat them as markup.

Thank you.
What I posted before was from the “rtp set debug on”
The full log reads out a lot of information and I couldn’t decide what is the relevant part for my problem. Any comments on that?

Either you are not receiving RTP in either direction, or RTP: logging doesn’t go to the console, as there is no RTP logging in your log above.

From the full log,you want the complete INVITE request, both inbound and outbound, and the complete contents of all responses to them and the associated ACK’s. If you are not famililare with how SIP works, I’d suggest reading up on that.

The reason for the full log is that information may be missing from the console. Time stamps may be missing, and people often don’t include far enough back, or even miss out the INVITE completely. It also guarantees that one will get other contextual information.

If it is extremely large, it could be that you are using FreePBX, in which case you really need to be on their forum, as people there are better able to wade through 100s of lines of their dialplan, or you are too open to the internet, and are being swamped by toll fraud attempts.

Thank you for the reply. So the INVITE and respective response is as follows. We see that RTP is occuring between both direction:

<--- Transmitting SIP request (1089 bytes) to UDP:192.168.200.210:5060 --->                                             
INVITE sip:24@192.168.200.210:5060;ob SIP/2.0                                                                           
Via: SIP/2.0/UDP 192.168.200.20:5060;rport;branch=z9hG4bKPjafaa57aa-8cc2-433a-bcb2-90c9c4d7c965                         
From: "23" <sip:23@192.168.200.20>;tag=b26cfb8d-044d-41a0-9201-48e10b248794                                             
To: <sip:24@192.168.200.210;ob>                                                                                        
Contact: <sip:asterisk@192.168.200.20:5060>                                                                             
Call-ID: 2ba1b5b7-c355-43dd-8d6e-4982088ebf31                                                                           
CSeq: 5161 
INVITE                                                                                                      
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, 
REFER           
Supported: 100rel, timer, replaces, norefersub                                                                          
Session-Expires: 1800                                                                                                   
Min-SE: 90                                                                                                              
P-Asserted-Identity: "23" <sip:23@192.168.200.20>                                                                       
Max-Forwards: 70                                                                                                        
User-Agent: FPBX-15.0.16.75(16.13.0)                                                                                    
Content-Type: application/sdp                                                                                           
Content-Length:   367                                                                                                                                                                                                                           
v=0                                                                                                                     
o=- 2044136647 
2044136647 IN IP4 192.168.200.20                                                                         
s=Asterisk                                                                                                              
c=IN IP4 192.168.200.20                                                                                                
 t=0 0                                                                                                                   
m=audio 15596 RTP/AVP 0 8 3 111 9 4 101                                                                                 
a=rtpmap:0 PCMU/8000                                                                                                    
a=rtpmap:8 
PCMA/8000                                                                                                    
a=rtpmap:3 GSM/8000                                                                                                     
a=rtpmap:111 G726-32/8000                                                                                               
a=rtpmap:9 
G722/8000                                                                                                    
a=rtpmap:4 G723/8000                                                                                                    
a=rtpmap:101 telephone-event/8000                                                                                       
a=fmtp:101 
0-16                                                                                                         
a=ptime:20                                                                                                              
a=maxptime:150                                                                                                          
a=sendrecv                                                                                                                                                                                                                                      
<--- Transmitting SIP request (1089 bytes) to UDP:192.168.200.210:5060 --->                                             
INVITE sip:24@192.168.200.210:5060;ob SIP/2.0                                                                           
Via: SIP/2.0/UDP 192.168.200.20:5060;rport;branch=z9hG4bKPjafaa57aa-8cc2-433a-bcb2-90c9c4d7c965                         
From: "23" <sip:23@192.168.200.20>;tag=b26cfb8d-044d-41a0-9201-48e10b248794                                             
To: <sip:24@192.168.200.210;ob>                                                                                         
Contact: <sip:asterisk@192.168.200.20:5060>                                                                             
Call-ID: 2ba1b5b7-c355-43dd-8d6e-4982088ebf31                                                                          
CSeq: 5161 
INVITE                                                                                                       
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, 
REFER           
Supported: 100rel, timer, replaces, norefersub                                                                          
Session-Expires: 1800                                                                                                   
Min-SE: 90                                                                                                              
P-Asserted-Identity: "23" <sip:23@192.168.200.20>                                                                      
Max-Forwards: 70                                                                                                        
User-Agent: FPBX-15.0.16.75(16.13.0)                                                                                    
Content-Type: application/sdp                                                                                           
Content-Length:   367                                                                                                                                                                                                                           
v=0                                                                                                                     
o=- 2044136647 
2044136647 IN IP4 192.168.200.20                                                                         
s=Asterisk                                                                                                              
c=IN IP4 192.168.200.20                                                                                                
 t=0 0                                                                                                                   
m=audio 15596 RTP/AVP 0 8 3 111 9 4 101                                                                                 
a=rtpmap:0 PCMU/8000                                                                                                    
a=rtpmap:8 
PCMA/8000                                                                                                    
a=rtpmap:3 GSM/8000                                                                                                     
a=rtpmap:111 G726-32/8000                                                                                               
a=rtpmap:9 
G722/8000                                                                                                    
a=rtpmap:4 G723/8000                                                                                                    
a=rtpmap:101 telephone-event/8000                                                                                       
a=fmtp:101 
0-16                                                                                                         
a=ptime:20                                                                                                              
a=maxptime:150                                                                                                          
a=sendrecv                                                                                                                                                                                                                                      
<--- Received SIP response (344 bytes) from UDP:192.168.200.210:5060 --->                                               
SIP/2.0 100 Trying                                                                                                      
Via: SIP/2.0/UDP 
192.168.200.20:5060;rport=5060;received=192.168.200.20;branch=z9hG4bKPjafaa57aa-8cc2-433a-
bcb2-90c9c4d7c965                                                                                                                    
Call-ID: 2ba1b5b7-c355-43dd-8d6e-4982088ebf31                                                                           
From: "23" 
<sip:23@192.168.200.20>;tag=b26cfb8d-044d-41a0-9201-48e10b248794                                             
To: <sip:24@192.168.200.210;ob>                                                                                         
CSeq: 5161 
INVITE                                                                                                       
Content-Length:  0                                                                                                                                                                                                                                                                                                                                                      
<--- Received SIP response (976 bytes) from UDP:192.168.200.210:5060 --->                                               
SIP/2.0 200 OK                                                                                                          
Via: SIP/2.0/UDP 
192.168.200.20:5060;rport=5060;received=192.168.200.20;branch=z9hG4bKPjafaa57aa-8cc2-433a-
bcb2-90c9c4d7c965                                                                                                                    
Call-ID: 2ba1b5b7-c355-43dd-8d6e-4982088ebf31                                                                           
From: "23" <sip:23@192.168.200.20>;tag=b26cfb8d-044d-41a0-9201-48e10b248794                                             
To: <sip:24@192.168.200.210;ob>;tag=JR25tZ34RyR254W2b8f0dGp39KA1qI82                                                    
CSeq: 5161 INVITE                                                                                                       
Contact: <sip:24@192.168.200.210:5060;ob>                                                                               
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS                        
Supported: replaces, 100rel, timer, norefersub                                                                          
Session-Expires: 1800;refresher=uac                                                                                     
Require: timer                                                                                                          
Content-Type: application/sdp                                                                                           
Content-Length:   323                                                                                                                                                                                                                           
v=0                                                                                                                     
o=- 2208988831 
2208988832 IN IP4 192.168.200.210                                                                        
s=pjmedia                                                                                                               
b=AS:84                                                                                                                 
t=0 0                                                                                                                   
a=X-nat:0                                                                                                               
m=audio 4000 
RTP/AVP 0 101                                                                                             
 c=IN IP4 192.168.200.210                                                                                                
b=TIAS:64000                                                                                                            
a=rtcp:4001 IN IP4 
192.168.200.210                                                                                      
a=sendrecv                                                                                                              
a=rtpmap:0 PCMU/8000                                                                                                    
a=ssrc:17864176 cname:05c601bd044e01ec                                                                                  
a=rtpmap:101 telephone-event/8000                                                                                       
a=fmtp:101 0-16                                                                                                                                                                                                                                     
-- PJSIP/24-00000016 answered PJSIP/23-00000015                                                                            
> 0x73e24c10 -- Strict RTP learning after remote address set to: 192.168.200.210:4000                                   
> 0x73e35908 -- Strict RTP learning after remote address set to: 192.168.200.230:4008                            
<--- Transmitting SIP request (429 bytes) to UDP:192.168.200.210:5060 --->                                              
ACK sip:24@192.168.200.210:5060;ob SIP/2.0                                                                              
Via: SIP/2.0/UDP 192.168.200.20:5060;rport;branch=z9hG4bKPj7683c5e6-5f2a-41cf-b452-e50310bd29c3                         
From: "23" <sip:23@192.168.200.20>;tag=b26cfb8d-044d-41a0-9201-48e10b248794                                             
To: <sip:24@192.168.200.210;ob>;tag=JR25tZ34RyR254W2b8f0dGp39KA1qI82                                                    
Call-ID: 2ba1b5b7-c355-43dd-8d6e-4982088ebf31                                                                           
CSeq: 5161 ACK                                                                                                          
Max-Forwards: 70                                                                                                        
User-Agent: FPBX-15.0.16.75(16.13.0)                                                                                   
Content-Length:  0                                                                                                                                                                                                                                                                                                                                                      
<--- Transmitting SIP response (995 bytes) to UDP:192.168.200.230:5060 --->                                             
SIP/2.0 200 OK                                                                                                         
Via: SIP/2.0/UDP 192.168.200.230:5060;rport=5060;received=192.168.200.230;branch=z9hG4bKPjW4K0b2P0eiE1BzT
5WJk1aSl0Dfg33C44                                                                                                                      
Call-ID: lQ857ke2Ne43pQq0h.q17NA2QA-4vo80                                                                               
From: <sip:23@192.168.200.20>;tag=T3N3R4c1tZ40UQ42x.t2c895aeT1DvZ1                                                      
To: <sip:24@192.168.200.20>;tag=b466589e-0db0-4d47-b549-d831a41b27ee                                                    
CSeq: 18764 INVITE                                                                                                      
Server: FPBX-15.0.16.75(16.13.0)                                                                                        
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, 
MESSAGE, REFER           Contact: <sip:192.168.200.20:5060>                                                                                      
Supported: 100rel, timer, replaces, norefersub                                                                         
Session-Expires: 1800;refresher=uac                                                                                     
Require: timer                                                                                                          
P-Asserted-Identity: "24" <sip:24@192.168.200.20>                                                                       
Content-Type: application/sdp                                                                                           
Content-Length:   265                                                                                                                                                                                                                           
v=0                                                                                                                     
o=- 2208988965 
2208988967 IN IP4 192.168.200.20                                                                         
s=Asterisk                                                                                                              
c=IN IP4 192.168.200.20                                                                                                 
t=0 0                                                                                                                   
m=audio 14366 RTP/AVP 0 8 120                                                                                           
a=rtpmap:0 
PCMU/8000                                                                                                    
a=rtpmap:8 PCMA/8000                                                                                                    
a=rtpmap:120 telephone-event/8000                                                                                       
a=fmtp:120 
0-16                                                                                                         
a=ptime:20                                                                                                              
a=maxptime:150                                                                                                          
a=sendrecv                                                                                                                                                                                                                                          
-- Channel PJSIP/24-00000016 joined 'simple_bridge' basic-bridge <5368ee44-f377-4608-aec8-
4bd9a0526afa>                 
-- Channel PJSIP/23-00000015 joined 'simple_bridge' basic-bridge 

<5368ee44-f377-4608-aec8-4bd9a0526afa>                    
> 0x73e24c10 -- Strict RTP switching to RTP 
target address 192.168.200.210:4000 as source                        
Got  RTP packet from    192.168.200.210:4000 (type 00, seq 017983, ts 000160, len 000160)                               
Got  RTP packet from    192.168.200.210:4000 (type 00, seq 017984, ts 000320, len 000160)                               
Got  RTP packet from    192.168.200.210:4000 (type 00, seq 017985, ts 000480, len 000160)                               
Sent RTP packet to      192.168.200.230:4008 (type 00, seq 009755, ts 000160, len 000160)                               
Sent RTP packet to      192.168.200.230:4008 (type 00, seq 009756, ts 000320, len 000160)                               
Sent RTP packet to      192.168.200.230:4008 (type 00, seq 009757, ts 000480, len 000160)                               
<--- Received SIP request (365 bytes) from UDP:192.168.200.230:5060 --->                                                
ACK sip:192.168.200.20:5060 SIP/2.0                                                                                     
Via: SIP/2.0/UDP 192.168.200.230:5060;rport;branch=z9hG4bKPjrKM19wA3K5d4D-y3njI1bTr1SdU4lSd3                            
Max-Forwards: 70                                                                                                        
From: sip:23@192.168.200.20;tag=T3N3R4c1tZ40UQ42x.t2c895aeT1DvZ1                                                        
To: sip:24@192.168.200.20;tag=b466589e-0db0-4d47-b549-d831a41b27ee                                                      
Call-ID: lQ857ke2Ne43pQq0h.q17NA2QA-4vo80                                                                               
CSeq: 18764 ACK                                                                                                        
Content-Length: 0                                                                                                                                                                                                                                                                                                                                                      
Got  RTP packet from    192.168.200.210:4000 (type 00, seq 017986, ts 000640, len 000160)                               
Got  RTP packet from    192.168.200.210:4000 (type 00, seq 017987, ts 000800, len 000160)                               
Got  RTP packet from    192.168.200.210:4000 (type 00, seq 017988, ts 000960, len 000160)                               
Sent RTP packet to      192.168.200.230:4008 (type 00, seq 009758, ts 000640, len 000160)                               
Sent RTP packet to      192.168.200.230:4008 (type 00, seq 009759, ts 000800, len 000160)                               
Sent RTP packet to      192.168.200.230:4008 (type 00, seq 009760, ts 000960, len 000160)                               
<--- Received SIP request (868 bytes) from UDP:192.168.200.230:5060 --->                                                
UPDATE sip:192.168.200.20:5060 SIP/2.0                                                                                  
Via: SIP/2.0/UDP 
192.168.200.230:5060;rport;branch=z9hG4bKPjh7U4HxT0AKD5oB53nq74y3A4D8x21kH3                            
Max-Forwards: 70                   
                                                                                     
From: sip:23@192.168.200.20;tag=T3N3R4c1tZ40UQ42x.t2c895aeT1DvZ1                                                        
To: sip:24@192.168.200.20;tag=b466589e-0db0-4d47-b549-d831a41b27ee                                                      
Contact: <sip:23@192.168.200.230:5060;ob>                                                                               
Call-ID: lQ857ke2Ne43pQq0h.q17NA2QA-4vo80                                                                               
CSeq: 18765 UPDATE                                                                                                      
Supported: replaces, 100rel, timer, norefersub                                                                         
Session-Expires:1800;refresher=uac                                                                                     
Min-SE: 90                                                                                                              
Content-Type: application/sdp                                                                                           
Content-Length:   323                                                                                                                                                                                                                           
v=0                                                                                                                     
o=- 2208988965 
2208988966 IN IP4 192.168.200.230                                                                        
s=pjmedia                                                                                                               
b=AS:84                                                                                                                 
t=0 0                                                                                                                   
a=X-nat:0                                                                                                               
m=audio 4008 
RTP/AVP 0 120                                                                                              
c=IN IP4 192.168.200.230                                                                                                
b=TIAS:64000                                                                                                            
a=rtcp:4009 IN IP4 
192.168.200.230                                                                                      
a=ssrc:42883853 
cname:0606457c023f695c                                                                                  
a=rtpmap:0 PCMU/8000                                                                                                    
a=rtpmap:120 telephone-event/8000                                                                                       
a=fmtp:120 0-16                                                                                                         
a=sendrecv                                                                                                                                                                                                                                             
> 0x73e35908 -- Strict RTP learning after remote address set to: 192.168.200.230:4008                            
<--- Transmitting SIP response (920 bytes) to UDP:192.168.200.230:5060 --->                                             
SIP/2.0 200 OK                                                                                                          
Via: SIP/2.0/UDP 
192.168.200.230:5060;rport=5060;received=192.168.200.230;branch=z9hG4bKPjh7U4HxT0AKD5oB5
3nq74y3A4D8x21kH3                                                                                                                      
Call-ID: lQ857ke2Ne43pQq0h.q17NA2QA-4vo80                                                                               
From: <sip:23@192.168.200.20>;tag=T3N3R4c1tZ40UQ42x.t2c895aeT1DvZ1                                                      
To: <sip:24@192.168.200.20>;tag=b466589e-0db0-4d47-b549-d831a41b27ee                                                    
CSeq: 18765 UPDATE                                                                                                      
Session-Expires: 1800;refresher=uac                                                                                     
Require: timer                                                                                                          
Contact: <sip:192.168.200.20:5060>                                                                                      
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, 
PRACK, MESSAGE, REFER           Supported: 100rel, timer, replaces, norefersub                                                                          
Server: FPBX-15.0.16.75(16.13.0)                                                                                       
Content-Type: application/sdp                                                                                           
Content-Length:   241                                                                                                                                                                                                                           
v=0                                                                                                                     
o=- 2208988965 
2208988968 IN IP4 192.168.200.20                                                                         
s=Asterisk                                                                                                              
c=IN IP4 192.168.200.20                                                                                                 
t=0 0                                                                                                                   
m=audio 14366 RTP/AVP 0 120                                                                                             
a=rtpmap:0 
PCMU/8000                                                                                                    
a=rtpmap:120 telephone-event/8000                                                                                       
a=fmtp:120 0-16                                                                                                         
a=ptime:20                                                                                                              
a=maxptime:150                                                                                                          
a=sendrecv                                                                                                                                                                                                                                             
> 0x73e35908 -- Strict RTP switching to RTP target address 192.168.200.230:4008 as source                        
Got  RTP packet from    192.168.200.230:4008 (type 00, seq 026017, ts 000160, len 000160)                               
Got  RTP packet from    192.168.200.230:4008 (type 00, seq 026018, ts 000320, len 000160)                               
Got  RTP packet from    192.168.200.230:4008 (type 00, seq 026019, ts 000480, len 000160)                               
Sent RTP packet to      192.168.200.210:4000 (type 00, seq 008271, ts 000160, len 000160)                               
Sent RTP packet to      192.168.200.210:4000 (type 00, seq 008272, ts 000320, len 000160)                               
Sent RTP packet to      192.168.200.210:4000 (type 00, seq 008273, ts 000480, len 000160)                               
Got  RTP packet from    192.168.200.210:4000 (type 00, seq 017989, ts 001120, len 000160)                               
Sent RTP packet to      192.168.200.230:4008 (type 00, seq 009761, ts 001120, len 000160)  
Got  RTP packet from    192.168.200.210:4000 (type 00, seq 017990, ts 001280, len 000160)

You are using FreePBX. Please note that suggestions here may be difficult or impossible to implement in FreePBX.

You are missing the in initial INVITE from 192.168.200.230

The RTP debug shows RTP both ways, so any RTP failure is outbound from Asterisk to the devices. My guess is that it is the result of an inbound firewall setting on the machine running the phones or a router close to them.

There doesn’t seem to be any NAT involved.

I guess I missed the initial part:

<--- Transmitting SIP response (558 bytes) to UDP:192.168.200.230:5060 --->                                             
SIP/2.0 401 Unauthorized                                                                                                

Via: SIP/2.0/UDP 
192.168.200.230:5060;rport=5060;received=192.168.200.230;branch=z9hG4bKPjy.O4Dzq4mdH4gu-
32vE6FBT00xG58ZC6                                                                                                                      

Call-ID: lQ857ke2Ne43pQq0h.q17NA2QA-4vo80                                                                               

From: <sip:23@192.168.200.20>;tag=T3N3R4c1tZ40UQ42x.t2c895aeT1DvZ1                                                      
To: <sip:24@192.168.200.20>;tag=z9hG4bKPjy.O4Dzq4mdH4gu-32vE6FBT00xG58ZC6                                               
CSeq: 18763 INVITE                                                                                                      

WWW-Authenticate: Digest 
realm="asterisk",nonce="1658476326/a67a721cb2086362317276e12d3c125d",opaque="5b2f34324b1
60d34",algorithm=md5,qop="auth"                                                                                                

Server: FPBX-15.0.16.75(16.13.0)                                                                                        

Content-Length:  0                                                                                                                                                                                                                                                                                                                                                      
<--- Received SIP request (368 bytes) from UDP:192.168.200.230:5060 --->                                                
ACK sip:24@192.168.200.20 SIP/2.0                                                                                       

Via: SIP/2.0/UDP 192.168.200.230:5060;rport;branch=z9hG4bKPjy.O4Dzq4mdH4gu-
32vE6FBT00xG58ZC6                            

Max-Forwards: 70                                                                                                        

From: sip:23@192.168.200.20;tag=T3N3R4c1tZ40UQ42x.t2c895aeT1DvZ1                                                        
To: sip:24@192.168.200.20;tag=z9hG4bKPjy.O4Dzq4mdH4gu-32vE6FBT00xG58ZC6                                                 
Call-ID: lQ857ke2Ne43pQq0h.q17NA2QA-4vo80                                                                               
CSeq: 18763 ACK                                                                                                        

Content-Length:  0                                                                                                                                                                                                                                                                                                                                                      
<--- Received SIP request (1223 bytes) from UDP:192.168.200.230:5060 --->                                               
INVITE sip:24@192.168.200.20 SIP/2.0                                                                                    

Via: SIP/2.0/UDP 
192.168.200.230:5060;rport;branch=z9hG4bKPjW4K0b2P0eiE1BzT5WJk1aSl0Dfg33C44                            
Max-Forwards: 70                                                                                                        
From: sip:23@192.168.200.20;tag=T3N3R4c1tZ40UQ42x.t2c895aeT1DvZ1                                                        
To: sip:24@192.168.200.20                                                                                               
Contact: <sip:23@192.168.200.230:5060;ob>                                                                               
Call-ID: lQ857ke2Ne43pQq0h.q17NA2QA-4vo80                                                                               
CSeq: 18764 INVITE                                                                                                      
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, 
MESSAGE, OPTIONS                        Supported: replaces, 100rel, timer, norefersub                                                                          
Session-Expires: 1800                                                                                                   
Min-SE: 90                                                                                                              

Authorization: Digest username="23", realm="asterisk", 
nonce="1658476326/a67a721cb2086362317276e12d3c125d", uri="sip:24@192.168.200.20", 
response="6ba83edcbd0241585c1fe31cf8daff96", algorithm=md5, 
cnonce="rvf105f4z515kdJ3Ap22qOe4zpm1sR41", opaque="5b2f34324b160d34", 
qop=auth, nc=00000001                                                                        
Content-Type: application/sdp                                                                                           
Content-Length:   347                                                                                                                                                                                                                           
v=0                                                                                                                     
o=- 2208988965 2208988965 IN IP4 192.168.200.230                                                                        
s=pjmedia                                                                                                               
b=AS:84                                                                                                                 
t=0 0                                                                                                                   
a=X-nat:0                                                                                                               
m=audio 4008 RTP/AVP 0 8 120                                                                                            
c=IN IP4 192.168.200.230                                                                                                

b=TIAS:64000                                                                                                            
a=rtcp:4009 IN IP4 192.168.200.230                                                                                      

a=sendrecv                                                                                                              

a=rtpmap:0 PCMU/8000                                                                                                    

a=rtpmap:8 PCMA/8000                                                                                                    

a=rtpmap:120 telephone-event/8000                                                                                       

a=fmtp:120 0-16                                                                                                        


a=ssrc:42883853 cname:0606457c023f695c                                                                                                                                                                                                            
== Setting global variable 'SIPDOMAIN' to '192.168.200.20'                                                            

<--- Transmitting SIP response (360 bytes) to UDP:192.168.200.230:5060 --->                                             
SIP/2.0 100 Trying                                                                                                      
Via: SIP/2.0/UDP 
192.168.200.230:5060;rport=5060;received=192.168.200.230;branch=z9hG4bKPjW4K0b2P0eiE1BzT
5WJk1aSl0Dfg33C44                                                                                                                      
Call-ID: lQ857ke2Ne43pQq0h.q17NA2QA-4vo80                                                                               
From: <sip:23@192.168.200.20>;tag=T3N3R4c1tZ40UQ42x.t2c895aeT1DvZ1                                                      
To: <sip:24@192.168.200.20>                                                                                             
CSeq: 18764 INVITE                                                                                                      
Server: FPBX-15.0.16.75(16.13.0)                                                                                        
Content-Length:  0  

If there is an inbound firewall that is causing the trouble, than it should behave similarly when I call from one peer to an account not registered/connected to the askterisk server (there the phone works fine). Or am i missing something?

Regarding FreePBX, I understood that it is just a UI to configure the Asterisk and it has no influence by itself on the call procedure. Am I correct?

The blockage appears to be outbound from Asterisk. The FreePBX firewalls wouldn’t block outbound traffic, so the assumption is that it is inbound to the peer. I think have interpreted inbound as relative to Asterisk. You’ll basically have to monitor at various point until you find where the RTP is blocked.

Not really. FreePBX is a toolkit for creating single tenant PABXes, based on abstractions like IVRs, routes, ring groups, which are not explicit constructs in Asterisk. Is is built on the concept of chaining blocks that create these abstractions, together, whereas Asterisk is based on a much more procedural model. Features in Asterisk, like extension pattern matching are only really exposed to power users, and within the constraints of the above framework.

It generally works by using a large amount of fixed code, in the form of dialplan macros (NB macros are deprecated in Asterisk), together with boiler plate code generated from the GUI. I think it also relies a lot on database lookups (performed by the macros).

It is very definitely not a development tool for arbitrary Asterisk configuration.

The large volume of standard code means that only the top tier of FreePBX power users (ones who understand both FreePBX and Asterisk) can really interpret the logs it produces.

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