cause=200; text="RTP Timeout" Allow: ACK,CANCEL,BYE,OPTIONS,INVITE Supported: outbound User-Agent: JsSIP 0.4.0-devel Content-Length: 0 opensdp.osdk.js:157 app call ended ["local", null, "RTP Timeout"] app.js:221 app call ended app.js:146 Fri Jul 11 2014 16:04:36 GMT+0400 ...
c=INIP4192.168.7.101t=00m=audio20902RTP/AVP08119897102103104105106101a=rtpmap:0PCMU/8000a=rtpmap:8PCMA/8000a=rtpmap:11L16/8000a=rtpmap:98iLBC/8000a=fmtp:98mode=20a=rtpmap:97iLBC/8000a=fmtp:97mode=30a=rtpmap:102SPEEX/8000a=rtpmap:103SPEEX/16000a=rtpmap:104SPEEX/32000a=rtpmap:105iSAC/...
SIP can work with specified protocols to complete session setup and media negotiation, such as Real-Time Transport Protocol (RTP), Real-Time Transport Control Protocol (RTCP), Session Description Protocol (SDP), Real-time Stream Protocol (RTSP), Domain Name System (DNS), Stream Control ...
Reason=SIP;\ cause=408;text="RequestTimeout">;index=1.1.1, <sip:User3@UA3.example.com?Reason=SIP; \ cause=487;text="Request Terminated">; index=1.1.2, <sip:User4@UA4.example.com?Reason=SIP;\ cause=603;text="Decline">; index=1.1.3 | | | | | | | /* Upon receipt of the...
Timestamp: 1677761066CSeq: 102 BYEReason: Q.850;cause=38P-RTP-Stat: PS=0,OS=0,PR=218,OR=34880,PL=0,JI=0,LA=0,DU=4Session-ID: bec855b57ab451688867fbbadf5caadd;remote=5333aeefa2415ed8ba456255a062bc5fContent-Length: 0 262380: *Mar 2 05:44:26.558: //13958/FA...
a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv *Oct 11 15:37:47.196: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentGTD: No GTD found in inbound container *Oct 11 15:37:47.196: //905/80BB30990800/SIP/...
NA13009832 OpenStage phone sends a SIP CANCEL request after receiving 200 OK from OSV leading to mixed RTP streams. NA13207788 wrong sip mobility state after failed MOB logoff / logon NA13201167 Security Log Upload Settings will be changed after time by itself NA13196426 There is not "Grou...
fix for sending ACK to 200 OK with wrong via branch and wrong to tag rtpwritefirst only for private ip's, not to servers add max call time and ring time options webcallme implementation MRTC 1.1 - Tuesday, June 14, 2016 fix webrtc gw incoming call ring issue ...
a=ptime:20 Unified Communications Manager also support other rtpmap attributes in addition to the CLEARMODE. It can identify X-CCD, CCD and G.nX64 rtpmap attributes as G.Clear codec in incoming SDPs. Unified CM supports sending one ...
CC_CAUSE_NO_ROUTE The called party cannot be reached because the network that the call has been routed through does not serve the desired destination. Call Setup Timeout Failure Typical scenarios: •No H.323 call proceeding. •No H.323 alerting or connect message received from the...