interview questionns inn globla logic

send invite , now call p forwarded request what error msg u wil get ?
ICMP error msg
408 request timeout
diff between tcp and udp
udp or tcp packet size is big ?
tcp
tcp handshake flow ?
syn ,syn-ack and ack
after establishing tcp connection , suppose if i lost connection , then what wil
happen ?
do i need to establish reconntectionn to transfer data ?
whethhher tx data happens from begining or from where it stopped ?
hw ur ip pbx works ? is it software or hardware ?
how u can achieve reliabliity usig udp ?
using Timers in invite and non invite state machines.
Normal basic call flow with proxy ?
explainn architecuter of ur SUT ?
wt is significannce of contact and via ?
how to cancel all the registrationns for a particular AOR ?
wt happens when max forward becomes zero ?
how the sdp offer works ?
if i did not send sdp in intital invite , how sdp exchange happens ?
if uac sends inv without sdp , in 200 response fromm uas , it gets sdp of uas
so it is the sdp offer , so while sending ack from uac, it sends final sdp and t
hen rtp flow starts
call park call flow?
rtp prot range ? can i use 5000 etc ?
wt are mandatory headers in sip methods ?
can BYE send directly to uas using contact without using proxy ?
telnet belongs to which layer ? RTP ,FTP etc ?

whhy udp and tcp ?

wher u use udp and where u use tcp ?
where MGCP come into picture ?
call flow of mgcp ?
wt is diff between mgcp and sip .
how ur sytem works ?
Which channel of E1 represends signalling ?
16,32 channel
Interview questions in ariccent gurgaon:
how DTMF Digits are transfered , when features like conf or transfer are sued
DTMF Digits are transfered in rtp packetes , which u can see in ethereal traces.
Diff b/w record - route and route headers
record route is set by proxies to stay in the path for future requests also.
once route set is created after receivinng 200 , route is created and used route
further requests withhin the dialog.
via is used to traverse the resposnse in the same path as that of request
Which headers wil change if ur usinng loose routing and strict routinng ? and
wt is the diff ?
loose and strict routings are idenntified bby the tag lr in record route header.
if lr is not present , it is strict routing which is defined by rfc 2543.
If a proxy is strict rouuting ,it expects its uri in the request uri header.
while forwardinng to next hop , thhe request uri is replaced with destination ho
p URI and forwards the reqest.
How SDP offer and ansewer works ?

UAC sends SDP parameters in inital innvite and

UAS sends SDP answer in 180 or 200 ok , mentioning its SDP parameters(like suppo
rted codecs g711 or g729) with its priorities.
If UAS wont support the codec's mentioned by UAC , then it responds with 488 no
t acceptable or 415 Un Supported media type.

Finally in ACK, UAC decides whcih codec to use and sends final SDP to UAS.
Suppose if UAC not sent SDP in initial invite , then the UAS sends its SDP off
er and UAC responds in ACK with its SDP response.
in this case , suppose if UAS wont support the codecs sent by UAS ,then first it
shoud respond with ACK and then it sends bye.
In UAC SDP if G711 and G729 are priorities and in UAS if it mentions the priori
ties are G729 and G711 , then the priortiy of UAC is chosen , ie G711.
SDP parameters can be exchaged any no of times , provided the offer and answer o
f SDP exist.
ie for instance
Invite is offer , 180 is anwer.
Now again UAC can send new SDP in Prack annd UAS responds its anwer in 200 OK fo
r Prack.
Again in 200 OK for inv can containn new SDP offer and ACK containns SDP answe
r model.
UPDATE also used for exchg of SDP parameters.

Payload types in RTP ?

If TCP is used as transport protocol for SIP , then SIP application which workss
in applicationn layer of OSI model is not required to take care about retransm
ission of sip messages as per State machine to achieve reliablity.
In this case , transport protocol TCP at transport layer takes care about reliab
ility by retransmitting messages at transport layer.
If sip with UDP as transport protocol uses retx of packets at application layer
using timers as reliabbility is not achieved by transport layer .
200 OK message is retx at SIP applicatin layer even if it uses TCP as transport
protocol.