when i am using "RTP relay on" then i cant receive packets.
i am getting all information from Active Session.
i am using
Payload=0(PCMU/8000)
packet size=172.
is there any problem with lishtening port?
what port i have to use at the receiver end?
the port which is in target or the send port...
is there any meaning of the taget port?
target: 192.168.x.x:21324
please do clear this problem
thanks
opu
RTP Relay
Moderator: Brekeke Support Team
RTP Relay
opu
-
- Posts: 528
- Joined: Tue Sep 20, 2005 9:10 am
- Location: Tannersville, Pennsylvania
Hello,
Let me help you understand the excellent tool Brekeke gave us by allowing us to see the session details. I'm sure you will find it most helpful in debugging your problem.
A call consists of 2 parts.
1) The call from the UA/Phone to the Brekeke sip server
2) The call from the Brekeke sip server to another UA or Provider
For example let's assume the call is from a sip phone to brekeke to a provider.
In the first leg of the call phone to provider you can see the session info.
The first leg is also broken into two parts. The first Listen/Send/Target refers to your Sip server so in my example we have:
Part 1
Listen-port = 10946
Send-Port = 10938
Target = SipServerIpAddress
This means that the Sip server is listening to your ip phone's voice stream(rtp) on port 10946, sending voice streams(rtp) to your ip phone from the sip server's port 10938 and the sip server ip address is the target.
Part 2
Listen-port = 10938
Send-Port = 10946
Target = IpPhoneIPAddress:Port
This is the inverse of part one but again it only deals with the sip servers ports, not the ip phone. The Target here does reveal the phones listening port as it follows the Target's ipaddress. This tells you that the phone wants it's rtp voice stream sent to that port. Typically, in a ip phone you can specify what range of ports the phone can use for rtp. In a Linksys spa-941 it defaults to 16384-16482. Now, what get's confusing is when the phone is behind a nat router and the sip server is not in that lan you may see a number for the ip phones port in the Target that you don't recognize. That is because the nat router changes it. If your phone is behind a nat router and your sip server is not in that lan and the target port for the ip phone is not what you expect then the router is changing it. This should still work since Brekeke sends it to the port specified in the target anyway. Some routers have the ability to exclude a port range from translation.
Post some more info about your situation or post your session details to get more detailed help.
Let me help you understand the excellent tool Brekeke gave us by allowing us to see the session details. I'm sure you will find it most helpful in debugging your problem.
A call consists of 2 parts.
1) The call from the UA/Phone to the Brekeke sip server
2) The call from the Brekeke sip server to another UA or Provider
For example let's assume the call is from a sip phone to brekeke to a provider.
In the first leg of the call phone to provider you can see the session info.
The first leg is also broken into two parts. The first Listen/Send/Target refers to your Sip server so in my example we have:
Part 1
Listen-port = 10946
Send-Port = 10938
Target = SipServerIpAddress
This means that the Sip server is listening to your ip phone's voice stream(rtp) on port 10946, sending voice streams(rtp) to your ip phone from the sip server's port 10938 and the sip server ip address is the target.
Part 2
Listen-port = 10938
Send-Port = 10946
Target = IpPhoneIPAddress:Port
This is the inverse of part one but again it only deals with the sip servers ports, not the ip phone. The Target here does reveal the phones listening port as it follows the Target's ipaddress. This tells you that the phone wants it's rtp voice stream sent to that port. Typically, in a ip phone you can specify what range of ports the phone can use for rtp. In a Linksys spa-941 it defaults to 16384-16482. Now, what get's confusing is when the phone is behind a nat router and the sip server is not in that lan you may see a number for the ip phones port in the Target that you don't recognize. That is because the nat router changes it. If your phone is behind a nat router and your sip server is not in that lan and the target port for the ip phone is not what you expect then the router is changing it. This should still work since Brekeke sends it to the port specified in the target anyway. Some routers have the ability to exclude a port range from translation.
Post some more info about your situation or post your session details to get more detailed help.
Thank you very much for your interest to help me.]
I am using my own code. for every call brekeke assign two port for send and listen e.g: 10000 and 10002.
before inviting i use two port in my media (m=audio xxxxx RTP/AVP ) of sdp(not dynamically).
e.g: for caller sdp its xxxxx and for callee sdp its yyyyy.
i observed that at the time of inviting when caller sdp is sent to callee its(xxxxx) changed into 10000 and when callee sdp is sent to callee its (yyyyy) changed into 10002.
that is callee gets a sdp containing port 10000 and caller gets a sdp containing port 10002.
/******
in active session i get
rtp-srcdst:
listen port: 10002
send port: 10000
rtp-dstsrc:
listen port: 10000
send port: 10002
seeing that i send rtp packet from caller in the port 10000
and from callee in the port 10002.
amazing thing is occurs in the target
for rtp-srcdst
target:192.168.0.2:yyyyy
rtp-dstsrc
target: 192.168.0.4:xxxxx
it should be mentioned that
caller is in 192.168.0.4
callee is in 192.168.0.2
and brekeke server is in 192.168.0.3
rtp packet is sent from both(caller and callee) to the ip 192.168.0.3(server pc)
i am using a LAN for testing.
problem is i cant receive any packet.
seeking a suggestion for the problem
Best Regards
I am using my own code. for every call brekeke assign two port for send and listen e.g: 10000 and 10002.
before inviting i use two port in my media (m=audio xxxxx RTP/AVP ) of sdp(not dynamically).
e.g: for caller sdp its xxxxx and for callee sdp its yyyyy.
i observed that at the time of inviting when caller sdp is sent to callee its(xxxxx) changed into 10000 and when callee sdp is sent to callee its (yyyyy) changed into 10002.
that is callee gets a sdp containing port 10000 and caller gets a sdp containing port 10002.
/******
in active session i get
rtp-srcdst:
listen port: 10002
send port: 10000
rtp-dstsrc:
listen port: 10000
send port: 10002
seeing that i send rtp packet from caller in the port 10000
and from callee in the port 10002.
amazing thing is occurs in the target
for rtp-srcdst
target:192.168.0.2:yyyyy
rtp-dstsrc
target: 192.168.0.4:xxxxx
it should be mentioned that
caller is in 192.168.0.4
callee is in 192.168.0.2
and brekeke server is in 192.168.0.3
rtp packet is sent from both(caller and callee) to the ip 192.168.0.3(server pc)
i am using a LAN for testing.
problem is i cant receive any packet.
seeking a suggestion for the problem
Best Regards
opu
Hi voipwell,
Thanks a lot. your post is really very helpful. and now i can receive packet from server.
If anyone face same type of problem i will recommend to see your post.
N.B. Now i have another problem i think it should discussed in different thread and hope i will get my solution.
Regards
Opu
Thanks a lot. your post is really very helpful. and now i can receive packet from server.
If anyone face same type of problem i will recommend to see your post.
N.B. Now i have another problem i think it should discussed in different thread and hope i will get my solution.
Regards
Opu
opu