rtp tos 808

2024-05-03


Am I missing something in the new asterisk version? Not sure if it is significant but I do see an error when looking in the logfiles. WARNING [10005] [C-00000005] chan_sip.c: Don't know how to indicate condition 36. I do see the call using RTP Audio and Video. 1879 [2022-08-10 11:56:13] VERBOSE[15449] netsock2.c: Using SIP RTP Audio TOS bits 184 .

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1. I have a 2.4 advantage ap and a 2.4 advantage sm. The SM has high priority turned on and the diffserv's are setup to defaults in each unit. Sipura ATA settings. SIP ToS/DiffServ Value:0x30 . RTP ToS/DiffServ Value:0x38. SIP CoS Value:3. RTP CoS Value:6. If I am doing things correctly. 0x30 is code point: 48. 0x38 is code point: 56.

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This protocol, RTP, was developed in the IETF and is in widespread use. The RTP standard actually defines a pair of protocols, RTP and the Real-time Transport Control Protocol (RTCP). The former is used for the exchange of multimedia data, while the latter is used to periodically send control information associated with a certain data flow.

For Asterisk Deployments in DiffServ environments, tos=0xB8 will set the Diffserv EF (Expedited Forwarding) bits. Asterisk 1.4. For DiffServ…to be completed… see doc/README.tos or doc/ip-tos.txt. tos_sip=cs3 ; Sets TOS for SIP packets. tos_audio=ef ; Sets TOS for RTP audio packets. tos_video=af41 ; Sets TOS for RTP video packets. Asterisk 1.6

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res_rtp_asterisk.c: DTLS failure occurred on RTP instance '0x7fe8c8024178' due to reason 'missing tmp ecdh key', terminating res_rtp_asterisk.c: RTP Read error: Unspecified. Hanging up. An earlier bug report listed this as a problem on FireFox : ASTERISK-25265 It is said to be fixed in 13.6

Beginner. 05-21-2021 12:27 AM. Greetings, I am having an issue with RTP Stream (I assume). So my setup is: ITSP->CUBE->CUCM->UCCX. This setup was working before and all of a sudden when I try calling from my mobile phone, the signaling works, the call gets connected but I get no audio whatsoever.

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