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Re: SIP Congestion
Posted: 05 Mar 2015 10:28
by haroun
32000 range
Re: SIP Congestion
Posted: 05 Mar 2015 11:11
by ranjithvkumar
Hi haroun,
Did you saw my trace ? any idea what may be the problem for no voice in incoming calls???

Re: SIP Congestion
Posted: 05 Mar 2015 12:16
by ranjithvkumar
In my
Outgoing:
Content-Type: application/sdp
v=0
o=- 1425574003 1425574003 IN IP4 10.201.20.45
s=SBC call
i=(o=IN IP4 10.80.10.154)
c=IN IP4 10.201.20.45
t=0 0
m=audio 24736 RTP/AVP 8 0 97
a=sendrecv
a=rtpmap:8 PCMA/8000
a=ptime:20
a=maxptime:30
a=rtpmap:0 PCMU/8000
a=ptime:20
a=maxptime:30
a=rtpmap:97 telephone-event/8000
Incoming:
Content-Type: application/sdp
v=0
o=- 1425573793 1425573793 IN IP4 10.201.20.45
s=SBC call
i=(o=IN IP4 10.80.10.10)
c=IN IP4 10.201.20.45
t=0 0
m=audio 23204 RTP/AVP 8 97
a=sendrecv
a=rtpmap:8 PCMA/8000
a=ptime:20
a=maxptime:30
a=rtpmap:97 telephone-event/8000
In my outgoing packets i am sending both PCMA & PCMU but in coming i am sending only PCMA. I need to add PCMU as my second option. See above highlighted.
So i enabled A-la to u-la conversion & also enabled multi-algorithrm support in system parameters. but still no change.
Can any suggest me where i can specify PCMU as my second option in incoming call packets.
Thanks in advance.
Posted: 05 Mar 2015 14:25
by cavagnaro
Incoming SDP is handled by caller but as he is offering a codec available for you it should work
Re: SIP Congestion
Posted: 06 Mar 2015 02:22
by ranjithvkumar
Any idea why incoming call voice not available ? what may be the problem ? Even i enabled accept both a-law & u-law but still i am sending only PCMA in incoming trace.
Re: SIP Congestion
Posted: 06 Mar 2015 04:21
by dryhouse
Hi,
With a wireshark trace you can see the ports that you send the rtp flow.
Is the RTP flow sincronous or asincronous?
Regards
Re: SIP Congestion
Posted: 06 Mar 2015 04:30
by ranjithvkumar
Hi dryhouse,
Now there is some change i noticed. When i establish the incoming calls, when i put the call on hold and retrieve the call then i can hear voice in both direction. and also after i establish the incoming call if i try to make new call and then if i resume the sip incoming call then also i can hear the voice in both direction. So is it because system is waiting for some activity to release the voice ?
Any idea ?
Posted: 06 Mar 2015 09:43
by cavagnaro
Check logs please. Answer has been given already. There is no magic answer for generic questions
Re: SIP Congestion
Posted: 08 Mar 2015 07:49
by haroun
pcma is ok for both path
but port changes from 32644 to 32514, yeah wirhesark can confirm witch ports are used fro rtp ans specially what happens when you do the manipulation put on hold and retreive the call changes will appear easly .
Re: SIP Congestion
Posted: 10 Mar 2015 04:26
by haroun
HI any update with wireshark
do you see g711 rtp or something else?