Questions tagged [rtp]

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20 views

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 ...
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1answer
162 views

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: ...
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1answer
23 views

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 ...
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1answer
362 views

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 ...
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0answers
161 views

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 = ...
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157 views

Asterisk/SIP - server with two NICs

We are trying to setup our Asterisk server with two NICs on it. One NIC is used to connect to a local network (which has access to internet and other sip trunks over internet) and other NIC is a ...
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1answer
699 views

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 ...
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1answer
340 views

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 ...
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2answers
627 views

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 ...
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0answers
100 views

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"...
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1answer
640 views

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 ?
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1answer
83 views

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 ...
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1answer
3k views

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 ...
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1answer
225 views

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 ...
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2answers
866 views

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 ...
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1answer
2k views

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. ...
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2answers
807 views

TCP/UDP Small Package Size optimization [closed]

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 ...
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1answer
4k views

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 - http://www.cisco.com/c/en/...
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0answers
501 views

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 ...
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0answers
280 views

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 ...
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2answers
561 views

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 ...
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1answer
920 views

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 ...
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1answer
61 views

Advice respect an architecture PBX [closed]

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 ...
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2answers
718 views

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 ...
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0answers
146 views

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 - ...
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1answer
912 views

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 ...
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1answer
1k views

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 ...
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0answers
732 views

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 ...