VoIP Mailing List Archives

[ehtasham.malik at expe...]

hello everyone, I am using sip.js for webRTC with Freeswitch, I am able to
initiate a call from sip.js to other sip extension but fail to answer the
incoming call on sip.js, and the following error occurs in the fs_cli.
2021-01-18 21:31:15.701411 [DEBUG] switch_rtp.c:4450 Starting timer [soft]
960 bytes per 20ms
2021-01-18 21:31:20.701423 [ERR] switch_core_media.c:9669 AUDIO RTP REPORTS
ERROR: [Remote Address Error!]
2021-01-18 21:31:20.701423 [NOTICE] switch_core_media.c:9670 Hangup
sofia/internal/5de16hg3@ccujprafrvr6.invalid [CS_CONSUME_MEDIA]
[INCOMPATIBLE_DESTINATION]
2021-01-18 21:31:20.701423 [ERR] sofia.c:8506 RTP Error!



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