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No audio using native android sip library - Stack Overflow
https://stackoverflow.com/questions/17436703/no-audio-using-native-android-sip-library
So I'm using the native sip library, and I can connect and register with the server just fine. And when I make the call, it hits a proxy that routes it to a regular phone call, then calls the number inputed. It will connect fine, and the phone on …
No audio when bridging two trunk calls - Asterisk SIP ...
https://community.asterisk.org/t/no-audio-when-bridging-two-trunk-calls/85714
Hi all, I am facing one weird issue with asterisk. I am using manager api to originate calls. I am successfully able to do that between one direct channel (on webrtc) and mobile number (through my sip provider) while everything works perfectly (able to hear voice, can end the call etc). But when I am trying to do the same between two mobile numbers (through …
SIP - No audio or one way audio :: Zoiper
https://www.zoiper.com/en/support/answer/for/android/11/SIP_-_No_audio_or_one_way_audio
SIP - No audio or one way audio ( on Android) « Back. When using SIP protocol one way or missing auido issues mostly appear due to configuration problems. The issues usually appear if the configuration is not adjusted accordingly to the wireless or 3g/4g network. In most cases this can be resolved by altering the account configuration. ...
Solved: incoming sip call with no audio drop after 10 ...
https://community.cisco.com/t5/ip-telephony-and-phones/incoming-sip-call-with-no-audio-drop-after-10-seconds/td-p/2158090
Hi all, i am facing a problem in sip line configuration. i am configuring sip line on branch router 2921. Incoming sip calls are disconnecting after 10 sec and there is no audio for either side. the other end is hearing only call progress tone even after my side answers the call. and out going ca...
No audio on Asterisk SIP call - Stack Overflow
https://stackoverflow.com/questions/5554837/no-audio-on-asterisk-sip-call
No audio on Asterisk SIP call. Ask Question Asked 10 years, 10 months ago. Active 1 year, 10 months ago. Viewed 34k times 9 I almost managed to init a 2 sided call (click to call): 1st to my office and the 2nd to my cell using Michal Niklas answer (thanks Michal) on Asterisk click to call. The major ISSUE is that the 2 call participants can not ...
No audio and no rtp traffic - Asterisk SIP - Asterisk ...
https://community.asterisk.org/t/no-audio-and-no-rtp-traffic/69420
– SIP/GO_VoIP_1-000000b9 answered SIP/VOIPTEST-000000b8 Audio is at 33636 Adding codec 100004 (alaw) to SDP Adding codec 100003 (ulaw) to SDP ... – Locally bridging SIP/VOIPTEST-000000b8 and SIP/GO_VoIP_1-000000b9 ... Also i found that in sip show registry no host is registered.
Audio Bridging: What It Is, Why You Need It - Omnitronics
https://www.omnitronicsworld.com/audio-bridging-what-it-is-why-you-need-it/
Audio Bridging provides a uniform way of interconnecting radio equipment from different manufacturers and in different frequency bands. Typically this is done at the repeater remote site to provide multiple paths within a single radio network. However, with advances in technology a vast array of additional applications are possible.
Native Bridging for iOS and Android in React Native | by ...
https://medium.com/simform-engineering/bridging-for-ios-and-android-in-react-native-64b8ce60a8c2
React Native Bridging is a concept that was developed by the React team to help the mobile app developers build their own custom modules, if not provided by the default Components given by React ...
"switching from simple_bridge technology to native_rtp"
https://asterisk-users.digium.narkive.com/H7310zSn/switching-from-simple-bridge-technology-to-native-rtp-issue
ast_clear_flag(&flags[0], SIP_REINVITE); The native RTP bridge in Asterisk 12 manages bridges between two RTP capable channels. The bridge can either be formed remotely (in which ... If you are getting one way audio when direct media is enabled, then one of the devices cannot find the other. This is most likely because
asterisk: IP address order may cause no audio · Issue …
https://github.com/irontec/ivozprovider/issues/511
then I have audio with rtpproxy on internal and external calls. In the first configuration -the one that has no audio on external calls- I can see how my sip device and my trunk provider are sending rtp packages to the correct requested rtp IP yyy.yyy.yyy.195 -the rtpproxy audio socket- but then rtpproxy sends those rtp packages to yyy.yyy.yyy.194.
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