Server Fault Stack Exchange
Server Fault Stack Exchange

Questions tagged [rtp]

30 questions
6
votes
2 answers

Causes of RTP jitter at the server

Investigating some call quality issues (0.5 – 1 second dead spots in calls) I took a packet capture of a phone call between two extensions on the same PBX. Since I was capturing from the PBX, I was rather surprised to see Wireshark reporting a huge…
miken32
  • 930
  • 1
  • 11
  • 32
4
votes
0 answers

RTP analysis - Discerning ptime (packetization time) for a given VoIP packet capture

I would like some help on the subject of an automated way of discerning the average packetization time (ptime) of a VoIP call's packet capture. The reason I am not depending on the value in the SDP is because some PBXs that I work with, send their…
bomp
  • 141
  • 2
3
votes
1 answer

Detect silence in RTP payload

I'm facing issue with a Asterisk-based ToIP infrastructure. Sometimes my phone received 'empty' RTP (payload entirely filled with 5d). I browsed RFC (3551, 3389) and can't find/understand relevant answers. I have found that two web pages that have…
Greeg
  • 31
  • 1
  • 2
2
votes
1 answer

When secure calling with asterisk (SRTP), why are client certificates needed for SIP devices?

I just added security to Asterisk by following this tutorial: https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial#SecureCallingTutorial-Keys Note that asterisk does not install by default with srtp by default. In order to be able to…
Tono Nam
  • 160
  • 1
  • 14
1
vote
0 answers

FreeSWITCH Dual Stack IPv4/IPv6

I'm currently trying to understand how I can enable my freeSWITCH to talk both IPv6 as well as IPv4. Currently, I thought it was going to be easiest to first create a set-up which works on IPv4 and then switch the IPv4 address for an IPv6…
Xabre
  • 143
  • 7
1
vote
1 answer

One Way Audio making "internal" SIP calls

I have just moved our phone system over to SIP using BT Voice Cloud and having a problem with one way audio when making "internal" calls. The original router (Vigor 2860n+) is the DHCP server, DNS server for the network and I need to keep that…
Gavin
  • 173
  • 1
  • 10
1
vote
1 answer

Asterisk uses the wrong variables from sip.conf

I completed this tutorial in order to make secure calls with asterisk. Secure Calling Tutorial | Asterisk Project Wiki I am running asterisk version 13.19.2 on Ubuntu version 16 (debian) and as soon as I added TLS and SRTP I ran into problems. Only…
Tono Nam
  • 160
  • 1
  • 14
1
vote
0 answers

rtpproxy is not listening at 7722

My retproxy is not listening on 7722 port .I have mentioned this in modparam("rtpproxy","rtpproxy_sock","udp:127.0.0.1:7722") modparam("rtpproxy","rtpproxy_sock","unix:/var/run/rtpproxy/rtpproxy.sock") also I have enable attributes on…
Avani badheka
  • 133
  • 7
1
vote
0 answers

Werbrtc2sip crashing intermittently. Aborted (core dumped)

I am having issues with making webrtc2sip actually work. Webrtc2sip is now getting crashed intermittently during a call. I have seen this even when only one call is active in a queue. By the way, my setup involves a single server with…
Kasinath Kottukkal
  • 211
  • 2
  • 3
1
vote
2 answers

RTP Stream is Visible from Internal IP but not from Public IP

I'm trying to stream an RTP stream to a port on an external ip. When I attempt to load the stream using the internal ip on the pc itself, everything works. When instead I try to stream using the external ip (even though I'm streaming to it), I…
Andrew
  • 121
  • 4
1
vote
1 answer

Security, hardening and NAT for RTP and SIP

I decided to create a VOIP server and it has gotten more complicated then I thought it would. So in order to make my server more secure, I keep it behind a pfSense appliance. I use IP filtering to reduce my online availability to only our remote…
user206106
  • 85
  • 1
  • 9
1
vote
2 answers

What determines the length of RTP packets?

I have established a SIP session between two clients. I observed the RTP trace between them on wireshark. The 'length' column has a value of 172 for the RTP packets that flow from Client 1 to Client 2 and has value 252 for packets that flow from…
user169253
1
vote
0 answers

Why adding capture filters breaks the traffic dump in wireshark / windump?

I have a strange issue while trying to capture RTP (UDP) traffic. I have a phone using IP 192.168.9.4 and a Windows 2003 PC connected to the same switch (actually to the monitor port of the switch - that's how I'm able to sniff the traffic). When I…
kyrisu
  • 491
  • 3
  • 5
1
vote
1 answer

Pbxnsip Music on Hold: Streaming shoutcast to RTP on Linux

So we have the PBXNsip PBX server. We want to hear shoutcast for the Music On Hold (MOH). They do offer an RTP for the MOH, but the music we want to hear is Shoutcast, and mp3 format. Is anyone experienced with using mplayer/ffmpeg/ffserver to…
Ryan
  • 39
  • 1
  • 6
1
vote
0 answers

H.323 to RTSP gateway?

Is there such a thing as a H.323 to RTSP gateway? Am I searching for the wrong terms? This site seems to imply that such a thing should already exists, but I cannot find anything at all. My end goal is to connect a Flash applet (via RTMP) on one end…
davr
  • 1,729
  • 3
  • 14
  • 24
1
2