How can I setup asterisk to dialog with sip devices using
the Multicast transmission protocol ?
Basically I have an asterisk box conected to a VSAT network.
On the other end I have a SIP box in a network receiving
the signal from the other VSAT.
The return from the SIP box to asterisk is unicast.
Is it possible to make it work ?
Asterisk SIP currently not support multicast.
Multicast rtp supported.
http://www.voip-info.org/wiki/view/Asterisk+MulticastRTP+channels
Related
In pktgen-dpdk is there any way to configure it to send and receive packets in the same port? for example port 0 send packets and receive it as it sends
#caetanocolinTorres simple answer is no special configure parameter in pktgen to do so.
explanation:
DPDK pktgen is build on top of DPDK library.
all the features of HW device is extracted by DPDK library.
there are some DPDK NIC which may support Internal loopback
to enable this one has to configure DPDK ports, which means edit PKTGEN source code and recompile.
Note: always share
OS version
dpdk version
NIC driver, firmware
original source if modified as the snippet.
I have some custom VPN code, written in C, which uses a Linux tun device.
Is it possible for this user-space code to see iptables marks that were attached to packets? Or do those marks not propagate as far as a tun interface?
We have a SIP trunk in our company. This SIP trunk is connected to Panasonic PBX and the PBX routes the calls to the extensions. Now we need a passive call recording server. The only task that this server should do is recording all the incoming and outgoing calls of the sip trunk and SHOULD NOT answer any call. So can we use asterisk as a recording server? If not what are other solutions?
If you can connect it like panasonic->e1 card->asterisk->incoming/outgoing sip or panasonic ->sip(if your panasonic have it)->asterisk->incoming/outgoing sip - then answer is yes.
If you want it use like panasonic recording card - answer is no.
I know about ISDN and its elements, but unable to apprehend ISDN card. What exactly is ISDN card and how does it connects to PSTN when it is installed in PC.
ISDN is digital protocol working via wire similar to ethernet one.
ISDN card for asterisk is just dumb interface card which read signal level over that wire and send it to software level - dahdi module. Dahdi analyze protocol and do all other magic, give asterisk rtp and isdn messages.
ISDN card connect with other "ISDN card" at provider side. After that provider connect it to other equipment in network, including PSTN gates.
ISDN is DIGITAL line, not analog like PSTN.
I'm developing software feature for a Session Boarder Controller(SBC).
I'm trying to establish a SIP call using two SIP clients and a Session Boarder Controller(SBC). Asterisk is used as the soft-switch.
When I call, the SIP signalling is working fine. But I am getting audio only in one direction. I captured rtp packets on all interfaces using wireshark. I observed that rtp packets in one direction is being dropped by asterisk.
Note: There is no send only attribute in any of the SIP/SDP messages.
I would like to know if there is any settings in asterisk that may cause this issue?
One more thing that I would like to know is that, from where a SIP client gets the RTP connection information. The port information is present in the media attribute
m=audio 16388 RTP/AVP 8 0 101
From where does the client get the transport IP address? is it from the "o=" field or "c=" field in the SDP or any other fields in the SDP or SIP?
You should troubleshoot the problem by capturing the complete call with Wireshark. Then look carefully at:
Client A initial INVITE: which port is it expecting media on (m= line) which address is it expecting media on (c= line)
SBC for Client A initial INVITE: If the SBC is anchoring the media (I assume so) check m / c lines
SBC for Client B initial INVITE: Which port / ip (m/c lines) is SBC for Client A expecting media on
Client B initial INVITE: Which port / ip (m/c lines) is SBC for Client B expecting media on
Are all nodes in this direction sending media on to the correct ports / ips (look at the RTP streams in wireshark)?
Then check the other direction (based on the SDP in the 183 or 200 (depending on your signaling flow)).
Note: In wireshark there is a nice feature which helps alot: Telephony --> VoIP Calls, which shows you the call flow more graphically