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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...
Netvanta 3120 inbound SIP calls lose audio after 30 ...
https://supportcommunity.adtran.com/t5/NetVanta-3100-Series/Netvanta-3120-inbound-SIP-calls-lose-audio-after-30-seconds/td-p/19437
1) Make sure that the SIP ALG is not enabled. This can be found in Data -> Firewall/ACL -> ALG Settings 2) In your VoIP port forwarding rule of the public interface make sure that Port Translation is not enabled. Data -> Security Zones -> Public -> "Your Port Forward Rule" -> Port Translation radial button to Disabled
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
On the 200 OK for the BYE message the SIP phone sends RTP stats, SCCP phone sends a ConnectionStatisticsRes message. This shows that the SCCP phone did not received the RTP stream from the SIP phone, this was lost on the network, this turned out to be a FW blocking the stream, this was fix by allowing the traffic on the FW.
How to resolve one-way or no-way audio on VoIP calls
http://info.teledynamics.com/blog/how-to-resolve-one-way-or-no-way-audio-on-voip-calls
The SIP server receives the initial communication and sets up a call control session between it and the external voice endpoint. In this call control session, the SIP server is informed of the destination endpoint. So, the SIP server communicates with the internal IP phone and causes it to ring. The user picks up and the call is connected.
Measuring VoIP Quality with SIP and RTP - Catchpoint
https://www.catchpoint.com/blog/voip-sip-rtp
Packet loss: Percentage of total packet loss during the audio session. Jitter TX/RX: Jitter is calculated based on the delay between packets that were expected to be delivered at a particular time, the metric is reported in milliseconds. Mean Opinion Score (MOS) TX/RX: MOS is a score given based on the quality of the audio session. This code ranges from 1 to 5, 1 being bad and …
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