Awesome and clean tutorial!! I am just tackling one way audio issues wirhmy asterisks etuo. Any tutorial hew to actually get rid of the issue once you locate the root cause? Thanks a lot!!!
how to identify if there no audio from both side i mean sender (caller) and callee(receiver) and this is happening for outbound external call only(When we call any germen number that scenario) give me some suggestion how i can find the root cause because i have checked from external SBC to Internal SBC no packet loss ,even no rule in firewall side ?
Isnot it that when rtp should be after sip invite, A to B, followed by sip 200, B to A, which means both way stream is expected? In case of one way issue, it happened on which way? What is the cause and how to solve it?
Appreciated I have received good info .....Audio issues are one which is very difficult to solve.....Any way we can solve the Audio issue or detect without RTP ... Because it's not adviceable to capture RTP every time :)
hi, thanks for the video very interest, but i have a question, hope you can get me a solution. this is about sip calls and rtp. i have filtered one call and i have more than one direction of rtp flows. im using wireshark version 3.4.4. when i play the rtp it plays all together. can i play one direction at a time. just like in this video. select the stream and play it in version 3.4.4..
great short video, would love to see more !
I guess I am kinda randomly asking but does anybody know of a good website to watch newly released series online?
@Ethan Creed i watch on FlixZone. You can find it by googling =)
Awesome and clean tutorial!! I am just tackling one way audio issues wirhmy asterisks etuo. Any tutorial hew to actually get rid of the issue once you locate the root cause? Thanks a lot!!!
how to identify if there no audio from both side i mean sender (caller) and callee(receiver) and this is happening for outbound external call only(When we call any germen number that scenario) give me some suggestion how i can find the root cause because i have checked from external SBC to Internal SBC no packet loss ,even no rule in firewall side ?
Isnot it that when rtp should be after sip invite, A to B, followed by sip 200, B to A, which means both way stream is expected? In case of one way issue, it happened on which way? What is the cause and how to solve it?
Great work!
Appreciated I have received good info .....Audio issues are one which is very difficult to solve.....Any way we can solve the Audio issue or detect without RTP ... Because it's not adviceable to capture RTP every time :)
hi, thanks for the video very interest, but i have a question, hope you can get me a solution. this is about sip calls and rtp. i have filtered one call and i have more than one direction of rtp flows. im using wireshark version 3.4.4. when i play the rtp it plays all together. can i play one direction at a time. just like in this video. select the stream and play it in version 3.4.4..
Superb...
very nice
thanks