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Asterisk SIP Call audio delay - Asterisk Support ...

    https://community.asterisk.org/t/asterisk-sip-call-audio-delay/30669
    When a caller calls in over a SIP trunk to our ITSP (Who’s using Metaswitch) the calls hit the IVR just fine, the caller can hear the IVR and can select options. However if they select an option to dial a ring group and we pickup there’s a 2-3 second delay in audio, the call shows connected, I can even see RTP flowing between the phone and the asterisk server but there’s …

Audio delay of 2s for incoming calls - "Answer() before ...

    https://community.asterisk.org/t/audio-delay-of-2s-for-incoming-calls-answer-before-dial-fixed-it/85711
    The Asterisk server is behind double NAT in a local LAN. The phones are in the same LAN. I’ve multiple sip accounts and sip providers. Problem: If I receive a call on one of my sip accounts, the audio from the external caller to the local phone is delayed by about 2s. Audio from local phone to external caller is not delayed!

how to delay sip 183 in asterisk - Stack Overflow

    https://stackoverflow.com/questions/15930747/how-to-delay-sip-183-in-asterisk
    The SIP 183 with SDP is indicating that my asterisk server is ready to send through audio and since there is no ringing within audio streams so, no ringback is observed. So, please tell me how to put delay in SIP 183. I am using asterisk 1.4 in centos 5

Asterisk Call Delay - Asterisk SIP - Asterisk Community

    https://community.asterisk.org/t/asterisk-call-delay/78920
    I have a strange issue when some calls have 10-15 seconds until Asterisk is sending INVITE to the peer. The problem is not permanently, few calls are working fine, and then 1-2 calls are with delay. For ex. one call with delay: CLI> pjsip show history 00003 1552335791 * <== 192.168.2.200:56089 INVITE sip:[email protected] SIP/2.0 00004 1552335791 * ==> …

Delay in processing of SDP - Asterisk Support - Asterisk ...

    https://community.asterisk.org/t/delay-in-processing-of-sdp/42052
    Three lines below you can see the “INVITE SDP” sent from our PBX (asterisk, 10.129.18.11) to the endphone (10.129.18.12). In this case there is more than 1 second of delay, but we experienced up to 5 seconds of delay. The version of …

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