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Is there anyway to test, by means Wireshark, a SIP connection calls quality in terms of: latency, jitter and Packet Loss? Thanks in advance

asked 28 Apr '14, 08:20

m4biz's gravatar image

m4biz
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accept rate: 0%


Is there anyway to test,

If to test means that Wireshark actively sends test traffic, then the answer is: No, because Wireshark is a passive network analysis/troubleshooting tool, thus it cannot send simulated traffic.

If to test means that you want to analyze the quality of a recorded RTP session, then the answer is: Yes

Telephony -> RTP -> Show all Streams

Regards
Kurt

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answered 28 Apr '14, 08:27

Kurt%20Knochner's gravatar image

Kurt Knochner ♦
24.8k1039237
accept rate: 15%

Hi Kurt. Thanks for your reply. I'd like to test a QoS solution (Untangle). For this, I'd like to see - during a SIP call placed by means an IP phone connected via Internet to an external SIP server (VoIP provider) - wich is the actual latency, jitter and packet loss. In particular I'd like to see if the VoIP call latency, jitter and packet loss increase when on the internal LAN there are download and uploads from other PCs (all connected to the same router/ADSL modem than IP phone). In other words I'd like to evaluate the effective "quality" of my QoS solution. I don't need that Wireshark generate any SIP/RTP traffic because the IP phone generate all traffic needed. I hope my question, now, is more understandable

(28 Apr '14, 08:53) m4biz
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question asked: 28 Apr '14, 08:20

question was seen: 6,646 times

last updated: 28 Apr '14, 14:11

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