Does RTP voice Calls will have 2 RTP sessions or only one

[nawaz.pasha]

I would like to know that the Audio RTP stream has 2 RTP streams for one call 2 way conversation...If someone is called and then the other person is also speaking from the other end does both will have 2 seperate RTP streams with different RTP channels...Please confirm I heard that it uses UDP is unidirection so does that mean the other person will initiate the other RTP seperate with its Source or uses the same RTP that is opened by the caller..

Since when we say 100mbps we are just saying one way direction in reality the throughput is 200mpbs. 100mbps for sending and 100mbps per second for receiving....this way we get 200mbps throughput for seconds we can upload/download...Please elaborate

Does this will have any bandwidth calculation....Also does the Region when we define does it even count the header bandwidth of Layer 2, 3 size for bandwidth or not...

[Vivek Batra · Accepted solution]

I would like to know that the Audio RTP stream has 2 RTP streams for one call 2 way conversation...If someone is called and then the other person is also speaking from the other end does both will have 2 seperate RTP streams with different RTP channels...Please confirm I heard that it uses UDP is unidirection so does that mean the other person will initiate the other RTP seperate with its Source or uses the same RTP that is opened by the caller..

UDP/RTP is bidirectional and both caller and called party will use separate RTP streams to send their voice. Hence in total for a call, you will be having two RTP streams. 

Both parties may use same different RTP ports on which they will be listen for RTP packets and in case of SIP, this information is negotiated in SDP during call setup.

Since when we say 100mbps we are just saying one way direction in reality the throughput is 200mpbs. 100mbps for sending and 100mbps per second for receiving....this way we get 200mbps throughput for seconds we can upload/download...Please elaborate

Bandwidth is mostly refereed in full duplex hence when you say 100 Mbps, you have 200 Mbps with you. Same is applicable while using various codecs for RTP. For e.g., G711 consumes 64 Kbps however you might say RTP is bidirectional so G711 must be using 128 Kbps.  But as you have already mentioned, bandwidth is mostly referred in full duplex hence when G711 is defined to use 64 Kbps, it's consuming bandwidth in both the direction (full duplex). 

Does this will have any bandwidth calculation....Also does the Region when we define does it even count the header bandwidth of Layer 2, 3 size for bandwidth or not...

​You never configure bandwidth under Regions. Bandwidth under Region is just a representation for group of codecs. For example, instead of only showing G711 and G722, bandwidth viz 64 Kbps is also showed there as just a TAG.

[nawaz.pasha]

Does that mean if G711 codec is using 32kbps for sending one channel of RTP(one way direction) + another 32kbps from other end and does combining the two uses 64kbps...is what you meant

I know bandwidth doesn't directly controlled through region but it does have the codec choice per call within or outside the region and its not the only mean to finalize the code but lot of other parameters are involved even the phones should support and codec prefereces matters...However, just wanted to know what about the Layer 2, 3 header is it part of the total bandwith per call or its separately has to be calcuclated when we design the network

[Vivek Batra]

Does that mean if G711 codec is using 32kbps for sending one channel of RTP(one way direction) + another 32kbps from other end and does combining the two uses 64kbps...is what you meant

I meant that in one direction, RTP payload uses 64 Kbps and same in other direction hence 128 Kbps in total however we refer bandwidth in full duplex only hence define G711 to consume 64 Kbps.

I know bandwidth doesn't directly controlled through region but it does have the codec choice per call within or outside the region and its not the only mean to finalize the code but lot of other parameters are involved even the phones should support and codec prefereces matters...

Correct

However, just wanted to know what about the Layer 2, 3 header is it part of the total bandwith per call or its separately has to be calcuclated when we design the network

Yes, we need to consider other overheads as well. Below is the nice document considering this aspect;

http://www.cisco.com/c/en/us/support/docs/voice/voice-quality/7934-bwidth-consume.html