![]() Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuriĪuthorization: Digest username=“5203”,realm=“asterisk”,nonce=“2ca50c27”,uri=“sip: :5060”,response=“3a543c63fbf4fc3280c47a4727ae5b40”,algorithm=MD5Ī=fmtp:106 maxplaybackrate=16000 sprop-maxcapturerate=16000 minptime=20 cbr=1 maxaveragebitrate=20000 useinbandfec=1įound audio description format speex for ID 110įound unknown media description format opus for ID 106įound audio description format PCMU for ID 0įound audio description format GSM for ID 3įound audio description format telephone-event for ID 101įound video description format H263-1998 for ID 115Ĭapabilities: us - (gsm|ulaw|g729|speex|speex16|h264|mpeg4|speex32), peer - audio=(gsm|ulaw|speex)/video=(h263p|g719)/text=(nothing), combined - (gsm|ulaw|speex) To: sip: :5060 tag=as72a1115bįrom: sip: :5060 tag=a8c1c110Īllow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE On an active call, when I press the video button on my Android, below SIP debug happens: I can make a call, but video won’t work. ![]() My two test SIP clients are both remote clients behind another firewall, one on Android, one on PC. My Asterisk is behind a firewall, SIP/RTP ports forwarded to it. ![]() I am trying to make video calls work over NAT. ![]()
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