For about 48 hours, I'm trying to play audio for who is calling me all by using NodeJS SIP and RTP directly.
When we calling I get the call on port 5060, and then I'm answering using the SIP/SDP specification. Then I'm start getting is voice on my RTP port. The issue is I'm unable to send voice back to him. What am I'm doing wrong?
I don't know yet how to build RTP packets, so I'm using the exact packets I received from the other side, and sent it back. I thought that will hear an echo. But he don't hear nothing.
He is using SIP and initiate a call. I was able to receive the call and record his audio.
The INVITE command
INVITE sip:1@104.154.78.142 SIP/2.0
Via: SIP/2.0/UDP 31.168.3.211:5060;branch=z9hG4bK071d7c26
Max-Forwards: 70
From: "0555042354" <sip:0523737233@31.168.3.211>;tag=as710f47f3
To: <sip:1@104.154.78.142>
Contact: <sip:0523737233@31.168.3.211:5060>
Call-ID: 36bde5b72f7a67956f7559a8581ea010@31.168.3.211:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.18.0
Date: Fri, 05 Feb 2021 00:22:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 236
v=0
o=root 2117083987 2117083987 IN IP4 31.168.3.211
s=Asterisk PBX 11.18.0
c=IN IP4 31.168.3.211
t=0 0
m=audio 16006 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
I took it's port (15574) from the INVITE command: m=audio 15574 RTP/AVP 8 101 and the IP also from the INVITE command: c=IN IP4 31.168.3.195 Now I'm sending back every UDP packet he sent me. But he don't hear nothing.
(It's after OK, and ACK by him).
ourSocket.on('message', (rtpPacketBuffer) => {
dgramServer.send(rtpPacketBuffer, 16006, 31.168.3.211)
})
In other words: According to RTP protocol, is sending back the packet can work, or do I need to reconstruct them?
(We're able to hear the other side, and record the call.)