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Solved: CUBE - One way audio - Cisco Community
https://community.cisco.com/t5/unified-communications/cube-one-way-audio/td-p/4432836
One way audio most of the time is a routing problem. So make sure that you don't have routing flapping causing intermittent RTP failure. Another common reason (which can be your case) is a firewall device inspecting your RTP/SCCP/SIP traffic.
Solved: CUBE: one way audio - Cisco Community
https://community.cisco.com/t5/ip-telephony-and-phones/cube-one-way-audio/td-p/2797681
Hello everyone, I'm having an issue with one way audio connecting to a sip provider. It looks like i'm not sending audio to them from the CUBE. here is my CUBE setup: version 15.4 service timestamps debug datetime msec localtime show-timezone service timestamps log datetime msec localtime show-ti...
Solved: CUBE One-Way Audio - Cisco Community
https://community.cisco.com/t5/collaboration-applications/cube-one-way-audio/td-p/4076969
CUBE One-Way Audio. Dears, hope you all doing well, We have this environment (call flow): ITSP (SIP Trunk) --> CUBE --> CUCM (5001 is AA) --> CUC (Customer Service 0 is a hunt group on CUCM extension number 1888) We have one way-audio and the call will disconnect after 10-15 seconds when calling to this hunt-group, the weird thing is when I do a "debug ccsip …
How to troubleshoot one-way / no audio issues - Cisco ...
https://community.cisco.com/t5/collaboration-voice-and-video/how-to-troubleshoot-one-way-no-audio-issues/ta-p/3164442
This example shows a one-way audio, the call flow is SIP phone calls an SCCP phone. SIP phone relevant info is marked in blue. SCCP phone relevant info is marked in orange. Since CUCM sends the correct IP address and port to each phone, this is …
How to punch SIP One Way Audio in the face - UCPros
https://ucpros.net/sip-one-way-audio/
we can go ahead and start punching one way audio in the face. Or in more civilized words, troubleshoot one-way audio. Troubleshooting one-way audio and no audio. The process includes three steps: 1. Find where RTP streams are sent to. 2. Verify whether RTP packets are received. 3. Work through possible issues. The logic is simple,
SIP Profiles on CUBE Enterprise Common Use Cases - Cisco
https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-border-element/118825-technote-sip-00.html
sip-profiles 1 inbound. One-way / No-way Audio Interoperability Issues with Provider voice class sip-profiles 200 request ANY sdp-header Audio-Attribute modify "a=inactive" "a=sendrecv" request ANY sdp-header Audio-Connection-Info modify "0.0.0.0" "CUBE’s IP" Remove the "UPDATE" Method Support to Avoid Interoperability Issues
Duplicate c= Lines in SDP Cause Intermittent One-way Audio ...
https://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-border-element/116010-dup-c-lines-problem-solution-00.pdf
This document provides a solution for intermittent one-way audio outbound calls over Session Initiation Protocol (SIP)/SIP Cisco Unified Border Element (CUBE) to various Internet Telephony Service Providers (ITSPs). Prerequisites Requirements Cisco recommends that you have knowledge of SIP. Components Used
Experiencing One Way Audio When Connecting via SIP
https://knowledgebase.paloaltonetworks.com/KCSArticleDetail?id=kA10g000000PLooCAG
Experiencing one-way audio when connecting via SIP (Session Initiation Protocol). Environment PAN-OS Cause SIP (Session Initiation Protocol) allows two endpoints to establish media sessions with each other. This is an application layer signaling protocol. The main signaling functions of the protocol are as follows: – Location of an end point.
Troubleshooting One Way Voice Issues - Cisco
https://www.cisco.com/c/en/us/support/docs/voice/voice-quality/5219-fix-1way-voice.html
Cisco Unified Communications Manager One-Way Audio Issue. This issue can only occur in an outgoing initial SIP call setup where MTP is required. In this case, the outgoing SIP INVITE message will contain an SDP offer. The issue may occur in these scenarios: Outgoing SIP trunk calls with Media Termination Point Required checked on the SIP trunk
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