B.Prathibha
B.Prathibha
INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/WSS 192.0.2.254;branch=z9hG4bK9756702 To: From: "prathibhab" ;tag=jud8opvn3c CSeq: 1 INVITE Call-ID: hs5dgk0n927dvgn1uaan Max-Forwards: 70 Contact: Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER Supported: outbound User-Agent: Browser Phone 0.3.20 (SIPJS - 0.20.0) Mozilla/5.0...
v=0 o=- 1381334577221906819 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE 0 1 a=extmap-allow-mixed a=msid-semantic: WMS f0cb88d3-b5f6-45ad-bf5e-2fc384cd5007 m=audio 63384 UDP/TLS/RTP/SAVPF 111 63 103 104 9 0 8 106 105 13...
Yes. With audio only this issue is not there.
In audio call, the following errors are displayed Attempt 1 to send STUN request to '142.250.25.127' timed out. Attempt 2 to send STUN request to '142.250.25.127' timed out. Attempt 3...
> So the reason for the error `PJMEDIA_SDP_EMISSINGRTPMAP`, is because the invite message is being truncated some how. > > Notice how the line: `m=video 63391 UDP/TLS/RTP/SAVPF 96 97 102...
there is no audio and video on both ends.
INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/WSS 192.0.2.212;branch=z9hG4bK6455491 To: From: "prathibhab" ;tag=k4uvq57tmc CSeq: 1 INVITE Call-ID: h6tmhv6n7utilvplvgk6 Max-Forwards: 70 Contact: Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER Supported: outbound User-Agent: Browser Phone 0.3.20 (SIPJS - 0.20.0) Mozilla/5.0...
ports 19302 and 3478 are opened in firewall. still getting the error: Attempt 3 to send STUN request to '216.93.246.18' timed out. Check that the server address is correct and...
SIP/2.0 400 Bad Request Via: SIP/2.0/WSS 192.0.2.229;rport=46718;received=10.10.10.70;branch=z9hG4bK9361963 Call-ID: d18fks6a4givifd2jm3p From: "prathibhab" sip:[email protected];tag=8155hus5kr To: sip:[email protected];tag=z9hG4bK9361963 CSeq: 2 INVITE Warning: 399 SIP "Missing SDP rtpmap for dynamic payload type (PJMEDIA_SDP_EMISSINGRTPMAP)" Server: Asterisk...
While on call:  After a minute: 