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@HengYongChao Thanks for your reply. The essence of this function "audio_element_set_uri" is to bootstrap to the URI, For example, "flash://tone/0_Bt_Reconnect. Mp3 ", how can I confirm this URI from my...
Yes,I have already generated my .bin file,but I don't know how to define my uri like "flash://tone/0_Bt_Reconnect.mp3". ------------------ 原始邮件 ------------------ 发件人: "espressif/esp-adf" ***@***.***>; 发送时间: 2022年3月25日(星期五) 下午4:27 ***@***.***>; ***@***.******@***.***>; 主题: Re: [espressif/esp-adf] When I ran the...
> i Thanks for your reply. Let two the same device with PJSIP in G.711 call each other.They can not establish dialog successfully. They have the same codec.
> > > i > > > Thanks for your reply. > > > Let two the same device with PJSIP in G.711 call each other.They can not establish dialog...
> @CAIJUN111 You only supply sip message. I mean you supply pjsip client's full log. @jimying oh,that the call's log(from register to invite) : (1251) VoIPDemo: [1.0] Initialize peripherals management...
> > @CAIJUN111 You only supply sip message. I mean you supply pjsip client's full log. > > @jimying oh,that the call's log(from register to invite) : > > (1251)...
@jimying I got the information according to the method you gave: > ```c > Unable to create media session > ```  00:00:07.344 pjsua_media.c ......Skipped updating media call00:1 (media...
> @CAIJUN111 Is the debugging code wrong? The printed SDPs (local/remote) are all wrong. I think at least remote-sdp should be right printed > > Maybe you can print local...
> In logs: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:+Z3/0O95f9G4symYXUmZ5TJ44gZgRAXXQAXHZQFs a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:MMhE4PUSf5q7qr6/Soc/D9yeHzcwJ/3ngjm6QvD1 > > must use SRTP? if use SRTP, need set "use_srtp" and "srtp_secure_signaling" when config account. @FredSE2021 thanks for your reply...
> I have met "488 Not Acceptable Here" before, when one site use SRTP, but other side not support SRTP. I solved the problem like that: one site to server,...