We have collected the most relevant information on Packet2packet Bridging Audio. Open the URLs, which are collected below, and you will find all the info you are interested in.


Getting Asterisk to Bridge Audio | MCB Systems

    https://www.mcbsys.com/blog/2008/11/getting-asterisk-to-bridge-audio/
    Packet2Packet Bridging = Audio is not going through the Asterisk core, it comes into the RTP stack and goes directly out. This decreases the amount of memory allocation that happens, and things require less processing.

General :: HOWTO RTP DTMF Troubleshooting : Bicom Systems

    https://support.bicomsystems.com/support/solutions/articles/67000668833-general-howto-rtp-dtmf-troubleshooting
    One can try to enable Packet2Packet bridging, but this will only work if certain conditions are met: 1. Both sides have to be SIP 2. Both sides have to have same audio codecs and same payload size (for instance 20ms) 3. Both sides have to have video support disabled 4. Both sides must not have options for DTMF processing (such as tT, instant ...

Tests of Asterisk Proxying Audio

    https://toao.net/VoIP/tests-of-asterisk-proxying-audio.html
    Local Bridge (Asterisk 11) / Packet2Packet Bridge (older Asterisk versions) Media is proxied by Asterisk, but relatively efficiently. The RTP comes in to the RTP stack and goes directly out without manipulation. Here is an example of what RTP debug looks like:

HOWTO RTP DTMF Troubleshooting - Bicom Systems Wiki

    https://wiki.bicomsystems.com/HOWTO_RTP_DTMF_Troubleshooting
    One can try to enable Packet2Packet bridging, but this will only work if certain conditions are met: 1. Both sides have to be SIP 2. Both sides have to have same audio codecs and same payload size (for instance 20ms) 3. Both sides have to have video support disabled 4.

Getting hold music after remote hangup on outgoing calls ...

    https://community.asterisk.org/t/getting-hold-music-after-remote-hangup-on-outgoing-calls/34995
    – Packet2Packet bridging SIP/6000-0000003a and SIP/204.xx.xx.xx-0000003b – Started music on hold, class ‘default’, on SIP/6000-0000003a == Spawn extension (internal, 1003, 1) exited non-zero on ‘SIP/6000-0000003a’ – Stopped music on hold on SIP/6000-0000003a localhostCLI> Any help with this would be greatly appreciated. Thanks.

What is Locally Bridging? : Asterisk

    https://www.reddit.com/r/Asterisk/comments/5936q6/what_is_locally_bridging/
    Thanks to RTP and bridging rewrites, the terminology is a bit different now (since about Asterisk 10, I believe). Now "native" and "Packet2Packet" bridges are both considered to be "native" bridges. The old "native" bridge is called a remote native bridge. The old "Packet2Packet" bridge is called a local native bridge.

Now you know Packet2packet Bridging Audio

Now that you know Packet2packet Bridging Audio, we suggest that you familiarize yourself with information on similar questions.