Server Fault Stack Exchange
Server Fault Stack Exchange

Questions tagged [rtp]

30 questions
0
votes
1 answer

IP Camera does not show open ports for RTP streaming but shows RTSP port open

I ran nmap on my IP camera to see what ports are open. I see a RTSP port open on port no 554, as expected. However, I do not see any ports open for RTP-RTCP streaming. Here is the output of nmap: PORT STATE SERVICE 22/tcp filtered…
asinix
  • 125
  • 5
0
votes
1 answer

Linux Headless Server Audio Player RTP stream

I am streaming audio from a linux server (192.168.0.10) to a headless client using ffmpeg. ffmpeg -i INPUT -acodec libmp3lame -ar 11025 --f rtp rtp://192.168.0.100:1234 On the headless client, I am trying to play the stream using vlc on the…
Menpress Malatae
  • 1
  • 7
0
votes
0 answers

What are RTP packets with zero timestamp and no payload?

I have an RTP traffics which some of the packets have zero timestamps and no payloads. Packet is : 10.. .... = Version(2) ..0. .... = Padding(false) ...0 .... = Extension(false) .... 0000 = Contributing Source ID count(0) 0... .... =…
saeed
  • 51
  • 3
0
votes
1 answer

RTP Proxy logs view

how to view rtp proxy logs separately . I actually want to view kamailio and rtp proxy logs differently.So I want to know that what are the command to view RTP proxy logs .can anyone help me ?
Avani badheka
  • 133
  • 7
0
votes
1 answer

Ideal way to monitor MOS scores with pfSense based routers

We currently have well over 200 pfSense routers in deployment at locations around the world running around 10,000 VoIP phones. Our issue is we need a way to monitor MOS and other QoS aspects and receive alerts should a threshold be reached. This…
Jason
  • 3,821
  • 17
  • 65
  • 106
0
votes
1 answer

VoIP: No audio, local IP in SIP header

I'm trying to get SIP trunking to work on my Bintec Elmeg hybird 130j with sipgate.de as trunking Provider. So far, call signalisation works fine, so I can call outside phones and also get called from outside my network. But there is no audio at…
NoMad
  • 302
  • 1
  • 4
  • 14
0
votes
2 answers

Does a SIP trunk relay the RTP stream or just set up the call?

Suppose I have a SIP PBX like Asterisk and a bunch of phones registered to it, and outgoing/incoming calls are handled through a SIP trunk. Do the RTP streams go directly between the phones and the SIP trunk provider or are they relayed through the…
mortabis
  • 64
  • 6
0
votes
1 answer

One-way audio problems with SIP over AT&T

I have a customer who has an VOIP PBX connected to a Level 3 fiber connection. He has offices all across the country using different ISPs. Two of those offices use AT&T, both in different states. One is T1, the other is DSL. For the past week, every…
pooter03
  • 436
  • 3
  • 11
0
votes
1 answer

Cisco ASA 5500 - SIP ports other than 5060

Is it correct that the SIP inspection in the ASA 5500 firewalls only kicks in for traffic on port 5060? There is some hint at this, while not 100% definitive, on Cisco Docs -…
nepdev
  • 381
  • 1
  • 6
  • 19
0
votes
1 answer

Advice respect an architecture PBX

i'm new on the world of PBX and network infrastructure. I worked in a model and wanted to know how feasible it is to implement it in a production environment. The requirement is to create a solution in the cloud to offer SME customers with our…
EmaX
  • 103
  • 2
0
votes
1 answer

SIP configuration with a Sonicwall firewall

I am working with a partner to enable SIP between a local SIP server and 2 Polycom phones located across a WAN connection. The server is located behind a Cicso ASA with SIP translation enabled, and the on the partner's side there is a Sonicwall…
chills42
  • 202
  • 4
  • 10
0
votes
0 answers

RTP audio Wrong sequence of packets

There is Freeswitch in VM bridged with the local machine. I have a call(say the current time) from baresip(console sip client) and browser(independently). They are on the same machine. Baresip calls and receives answers successfully. Wireshark…
Victor Shelepen
  • 101
0
votes
0 answers

No RTP engine was found. Do you have one loaded? Asterisk-18.10.1

I have been trying to install Asterisk-18.10.1 version on my ubuntu(20.04.4) running inside VM. I was able to maintain connection from GoTrunk SIP endpoint and Zoiper as softphone. Followed https://github.com/GoTrunk/asterisk-config/tree/dynamic-ip…
xenon
  • 1
0
votes
1 answer

Decoding TCP packets as RTP in Wireshark

I'm troubleshooting a WebRTC video calling problem in my app and i'm using Wireshark. One end of my video call is a web app running in my browser window and the other end is a Unity based app on an Android device. This is built with WebRTC. In…
Salbrox
  • 41
  • 1
  • 4
-1
votes
2 answers

TCP/UDP Small Package Size optimization

I currently have the problem, that when a LOT of RTP streams (>800) go to one server the max. speed is onlynabout 70-80 Mbit (on a Gigabit LAN - all Hardware components are Gigabit components) 130 byte payload - with large payload everything is…
pinas
  • 171
  • 1
  • 1
  • 5
1
2