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Found RTP audio format 103Found RTP audio format …
https://pastebin.com/zFepwdwN
Found RTP audio format 101 ... Found audio description format speex for ID 102 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x50120c (ulaw|alaw|speex|g722|h263p|mpeg4)/video=0x0 (nothing)/text ...
Found RTP audio format 0Found RTP audio format …
https://pastebin.com/JXFTmPq1
Found RTP audio format 0 Found RTP audio format 101 Found RTP audio format 100 Peer audio RTP is at port xxx.xxx.xxx.xxx:18756 Found audio description format telephone-event for ID 101 Found unknown media description format X-NSE for ID 100 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) ...
Skype connect sip profile to Asterisk - Microsoft Community
https://answers.microsoft.com/en-us/skype/forum/all/skype-connect-sip-profile-to-asterisk/a492986d-209f-45b4-9e8c-5ef00a7e6934
Found peer 'skype' for '3727120253' from 63.209.144.201:5060 //call from peer skype == Using SIP RTP CoS mark 5. Found RTP audio format 8. Found RTP audio format 0. Found RTP audio format 18. Found RTP audio format 13. Found RTP audio format 101. Found audio description format PCMA for ID 8 //trying to use these codecs formats
Process_sdp: Failing due to no acceptable offer found ...
https://community.asterisk.org/t/process-sdp-failing-due-to-no-acceptable-offer-found/90344
t=0 0 m=audio 13870 RTP/AVP 8 0 4 18 13 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000 a=fmtp:4 bitrate=6.3;annexa=yes a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:13 CN/8000 <-----> — (21 headers 13 lines) — Sending to z.z.z.z:5060 (no NAT) Sending to z.z.z.z:5060 (no NAT)
ios - Missing Audio/Video SDP issue - Stack Overflow
https://stackoverflow.com/questions/47656359/missing-audio-video-sdp-issue
Also I used Asterisk PBX 12.6.1 for VOIP server. I made audio call from A to B. I have confirmed SIP log from asterisk server and exactly Asterisk got audio SDP only. But when Invite Message sent by NAT to peer, I see the video SDP from Invite message. This mean, when Invite message passed NAT, video SDP added to Invite message.
SIP call fails with a SIP error 484 address incomplete ...
https://community.asterisk.org/t/sip-call-fails-with-a-sip-error-484-address-incomplete/35493
[Jun 6 07:17:47] VERBOSE[7881] chan_sip.c: Found RTP audio format 101 [Jun 6 07:17:47] VERBOSE[7881] chan_sip.c: Found audio description format PCMU for ID 0 [Jun 6 07:17:47] VERBOSE[7881] chan_sip.c: Found audio description format G726-32 for ID 2
'[asterisk-users] Found unknown media description …
https://marc.info/?l=asterisk-users&m=121845009729559
Content-Type: application/sdp Content-Length: 1123 P-hint: outbound v=0 o=- 1218448446 197568495 IN IP4 127.0.0.1 s=- c=IN IP4 192.168.0.176 t=0 0 a=ice-pwd:00gnam9Pd+SG6KzQNLf1fS a=ice-ufrag:Xng7 m=audio 19504 RTP/AVP 103 18 102 0 8 97 119 117 100 101 13 105 106 a=rtcp:19505 a=candidate:4 1 UDP 2122300927 192.168.0.176 …
Outbound Calls Dropping after 15 minutes - FreePBX ...
https://community.freepbx.org/t/outbound-calls-dropping-after-15-minutes/38029
useragent=FPBX-13.0.190.2(13.12.1) disallow=all allow=ulaw allow=alaw allow=gsm allow=g726 context=from-sip-external callerid=Unknown notifyringing=yes notifyhold=yes tos_sip=cs3 tos_audio=ef tos_video=af41 alwaysauthreject=yes limitonpeers=yes session-timers=originate session-expires=10800 session-minse=300 session-refresher=uas …
extension sip no se le escucha - Issabel
https://forum.issabel.org/d/718-extension-sip-no-se-le-escucha
t=0 0 m=audio 8016 RTP/AVP 3 0 8 110 98 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:110 speex/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <----->--- (14 headers 15 lines) ---Sending to 92.185.176.157:5061 (NAT) Sending to 92.185.176.157:5061 (NAT)
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