RTP Packets – to understand voice quality issues because RTP carries the voice packetsĪs you can see I have filtered SIP packets from a treasure of different packets which were flowing through my network.SIP Packets – to understand the call signalling mechanism.We need to understand two types of packets to get a proper understanding of a VoIP communication scenario: Here you will see all the basic SIP signalling parameters which we discussed above. Now let’s open a SIP trace in the Wireshark: this is simple VoIP call between two extensions configured on two local Asterisk PBXs: Once you will open it for the first time, you will see the following screen: Now let’s open Wireshark, you can download the latest version from:, I am using the 64-bit version. A particular SIP call is ended by sending a Bye message by either party, once a Bye message is sent the request will get stopped and RTP session dropped. After sending 1xx informational messages, Bob will send a 200 Ok message, which marks the establishment of a call and RTP stream is established after that instant. These SIP methods are used to convey different informational messages about status of a particular call. The numbers (100, 180 etc.) that you are seeing are known as SIP methods. When Alice initiate a SIP call, an INVITE packet is sent to Bob, in return Bob reply with 100 Trying or 180 Ringing message back to Alice. Suppose we have a two SIP devises Alice and Bob. SIP is one of the fundamental building blocks of today modern VoIP communication, not only used for voice communications, but also for multimedia session establishment, instant messaging or for some gaming session. Read Part 1 of Using Packet Analysis hereĪs we are working in the world of VoIP, we do need to know about the protocols that make this communication possible.
0 Comments
Leave a Reply. |
AuthorWrite something about yourself. No need to be fancy, just an overview. ArchivesCategories |