Asterisk can modify SIP packets to direct the caller and destination to establish an RTP session with itself, rather than with each other. It’s up to the endpoint and if NAT Is involved the router as to what source port is used for media originating from the endpoint.
I normally specify a range of 1000 ports, which is plenty. Es importante tener en mente que la comunicacion es bidereccional por lo tanto se deben abrir los puertos UDP 10000 a 20000 para trafico entrante y saliente, asi como el puerto UDP/TCP 5060, si hay un firewall de por medio en cada localidad, se deben configurar para permitir este trafico en cada una de las redes IP donde existan telefonos IP, de lo contrario no van a poder comunicarse.
It does not, and can not, control the remote endpoint. This month, the Asterisk project performed two security releases to address an unauthorized RTP data disclosure vulnerability in its real-time transport protocol (RTP) stack. I really need a SIP channel to RTP ports (at least the local one) mapping. The port number range is 10000 to 20000 by default, it can be changed in FreePBX, menu Settings – Asterisk SIP Settings, field RTP Port Ranges. You need 2 RTP ports per simultaneous call in or out.
Note: Zulu uses the same rtp port configuration as SIP.
This doesn't seem to be happening here. Safe to open this up to untrusted networks, as your RTP traffic can come from anywhere your Zulu users are connecting from. Used for handling media during a call The rtp.conf file configures the port range that Asterisk uses for its RTP ports. Enable ICE support; Tell Asterisk to send media across the same transport that we receive it from. My issue is that on the Asterisk side, I can dig and find a call in the logs and the CDRs, but I can't find a way to log the RTP ports of both call legs which were used for a specific call going through the Asterisk. It is my understanding that part of the SIP negotiation is sending traffic on ports that the receiving party has open. RTP ports. Specify which certificate files to use for TLS negotiations with this endpoint and our verification and setup methods.
From a debugging perspective, it's a nightmare. Media - RTP: The port can be changed by going to Settings → Asterisk SIP Settings → General SIP Settings Tab. This is useful in situations where two SIP clients may not have direct access to each other, most commonly, when one or both of the SIP clients are behind a NAT. These ports also have to be forwarded if remote extensions are desired. Opening 98% of the available ports seems massively unnecessary.
If calls can be made but either one way or no audio at all is experienced, forwarding RTP ports often helps. Overview. Notify Asterisk to expect the AVPF profile (secure RTP) Setup the DTLS method of media encryption.