I just can't get rtpengine to work. i have tried multiple configurations, but to no avail. I would appreciate any help/advice I can get. Note that calls work good if rtpengine is disabled.
Here is my setup =>
Public IP: 20.20.20.20
Private IP 10.10.10.10
flow =>
webrtc client <-> kamailio+rtpengine <-> asterisk <-> kamailio <-> legacy sip phone
rtpenngine startup (I have tired a few different startups) =>
rtpengine --interface=int/10.10.10.10 --interface=ext/10.10.10.10!20.20.20.20 --listen-ng=127.0.0.1:12221 --pidfile=/var/run/rtpengine --dtls-passive -f -m 10000 -M 20000 -E
kamailio =>
Invites: rtpengine_manage("trust-address replace-origin replace-session-connection direction=ext direction=int ICE=remove RTP/AVP");
Reply's: rtpengine_manage("trust-address replace-origin replace-session-connection ICE=force RTP/SAVPF");
I have tried direction ext ext; and many other combinations, each producing it's own incorrect behavior.
I would really appreciate any help!
Thank you,
Steve
Try using the actual interface name in your configuration and listen on the loopback and the internal IP.
--interface=eth1/10.10.10.10!20.20.20.20
--listen_ng=127.0.0.1:12221
--listen_ng=10.10.10.10:12221
Configure Kamailio to contact rtpengine on 10.10.10.10:12221.
Rtpengine modified packets can exceed the UDP size limit, therefore if you are using UDP for signaling anywhere, check if these packets are being fragmented as a result. If that is the case, then the only real solution is to use TCP for all legs.
Related
I need to exercise some hardware by sending a network traffic with it. While it is doing it I will probing some of the lines with an oscilloscope. Need to verify signaling. The problem is that I won't be able to connect to any server during the test. Many reasons for that, one of each is that hardware isn't complete yet.
Does anyone know if there is a away to generate network traffic with iperf without using a server? All I need is to just send some data, don't need to know if it was received. If there isn't can someone point me to a tool that can do that.
iperf UDP will do it you just need to make sure there is an arp entry for the destination (enter it manually) or use a multicast destination which doesn't require ARP, e.g. iperf -u -c 239.1.1.1 -b 10M
I am using csipsimple as sip client and asterisk server to set up call. Calls are made between 2 sip clients but voice is not getting transferred.
Calls are made between 2 sip clients using AMI.
I can give my asterisk cli log.
Can anybody please give me some idea to solve this issue?
Thanks
More info would be useful. First, make sure both clients are registered, and can use at least one common codec. In most cases, these aren't the problem. It's usually a NAT/Firewall issue. Are the two clients on the same subnet? Is there any firewall rules blocking the communication?
SIP signaling usually goes on udp:5060. But that seems working. Media is tricky. In each call, the ports for RTP audio changes, in the range specified in rtp.conf. This RTP traffic goes over UDP as well. By default it't 10000-20000.
If there is only routing done between the two endpoints, it should still be fine. NAT (Network Address Translation) is your main concern. Take a look at iptables, sip_nat_conntrack. To debug, use asterisk's sip set debug on command and look for the SIP headers and verify the correct IP addresses.
I have UDP network traffic arriving on my machine (OSC traffic from an iPad, to be exact) and I want two programs to be able to receive that traffic. The problem is that I can't bind to the same network port with two programs at once and I can't send to multiple ports with the iOS app I'm using. How can I solve this problem?
You can use the power of the command line for this. The following snippet uses socat (probably needs to be installed beforehand) and tee (should be preinstalled on any OS X or Linux).
socat -u UDP4-RECVFROM:8123,fork - | tee >(socat -u - UDP4-SENDTO:localhost:8223) | socat -u - UDP4-SENDTO:localhost:8323
Explanation: socat listens for traffic on UDP port 8123, pipes it to tee, which pipes it to two other instances of socat forwarding it to ports 8223 and 8323 on localhost respectively. With your two programs you need to listen to those ports on localhost.
While the answer with using socat is elegant it is not clear for me, what you are trying to do:
both programs should receive all parts of the traffic and they will only receive and not reply. This can be done with the proposed socat way
both program should receive all parts of the traffic and there reply will be mixed together (how?)
each of the programs should only receive parts of the traffic, e.g. the one which the other did not get. This should be possible if both of your programs use SO_REUSEADDR, SO_REUSEPORT. Replies will then be mixed together.
or do you actually want to communicate with each of the programs seperatly - then you would have to use either multiple sockets in the iOS app (which you don't want to do) or built your own protocol which does multiplexing, e.g. each message is prefixed with there target app and on the target machine a demultiplexer application will receive all packets and forward them to the appropriate application and wrap the replies back in the multiplexing protocol.
In summary: please describe the problem your are trying to solve, not only one small technical detail of it.
The problem is that I can't bind to the same network port with two programs at once
Yes you can. Just set SO_REUSEADDR and maybe SO_REUSEPORT on both of them before you bind.
I have problem with identifying communication established by TCP.
I have to identify first completed communication, for example first complete http communication.
I have dump .pcap file with capture. I know that communication should start by three way handshake ( SYN, SYN - ACK, ACK ) and then closing of communication by double FIN flag from both side.
But I have a lot of communication in that dump file.
So here is the question. Which things i need to remember to match exact one communication ?
I thought about source IP, destination IP, protocol, maybe port but i am not sure.
Thank you for every advice.
And sorry for my english.
You stated that you need:
To identify a particular conversation
To identify the first completed conversation
You can identify a particular TCP or UDP conversation by filtering for
the 5-tuple of the connection:
Source IP
Source Port
Destination IP
Destination Port
Transport (TCP or UDP)
As Shane mentioned, this is protocol dependent e.g. ICMP does not have the concept of
ports like TCP and UDP do.
A libpcap filter like the following would work for TCP and UDP:
tcp and host 1.1.1.1 and port 53523 and dst ip 1.1.1.2 and port 80
Apply it with tcpdump:
$ tcpdump -nnr myfile.pcap 'tcp and host 1.1.1.1 and port 53523 and dst ip 1.1.1.2 and port 80'
To identify the first completed connection you will have to follow the timestamps.
Using a tool like Bro to read a PCAP would yield the answer as it will list each connection
attempt seen (complete or incomplete):
$ bro -r myfile.pcap
$ bro-cut -d < conn.log | head -1
2014-03-14T10:00:09-0500 CPnl844qkZabYchIL7 1.1.1.1 57596 1.1.1.2 80 tcp http 0.271392 248 7775 SF F ShADadfF 14 1240 20 16606 (empty) US US
Use the flag data for TCP to judge whether there was a successful handshake and tear down.
For other protocols you can make judgements based on byte counts, sent and received.
Identifying the first completed communication is highly protocol specific. You are on the right track with your filters. If your protocol is a commonly used one there are plug ins called protocol analyzers and filters that can locate "conversations" for you from a pcap data stream. If you know approximate start time and end time that would help narrow it down too.
I've been using tcpdump for about a month now, and recently, it has stopped capturing any packets that were not sent to or from the computer running tcpdump. I've stripped down my command to just:
sudo tcpdump -i en2
I've checked my interfaces with ifconfig, and en2 is in "PROMISC" mode. When specifying a specific host as a filter, I only see a few "arp" messages but nothing compared to what is actually going on in the network.
Any ideas why this would be happening? Much appreciated if anyone can offer some advice!
Richard
ps, sorry for the re-post, i wanted to register this time! (new to s.o.)
Do you know what network equipment is used \ if there has been a change recently?
One possible explanation is that your computer is connected to a switch (and not a hub) the switch sends to your adapter only traffic intended to your MAC address, and broad casts (hence the ARP)
one way to check this is to send broad casts from other computers in the network (just use ping 255.255.255.255) and see if you can see anything.