SIP Congestion

haroun
Senior Member
Posts: 1454
Joined: 29 Mar 2010 11:09

Re: SIP Congestion

Post by haroun »

32000 range
ranjithvkumar

Re: SIP Congestion

Post by ranjithvkumar »

Hi haroun,

Did you saw my trace ? any idea what may be the problem for no voice in incoming calls??? :(
ranjithvkumar

Re: SIP Congestion

Post 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.
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cavagnaro
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Post by cavagnaro »

Incoming SDP is handled by caller but as he is offering a codec available for you it should work
ranjithvkumar

Re: SIP Congestion

Post 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.
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dryhouse
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Posts: 165
Joined: 06 Apr 2010 06:11
Location: Madrid,Spain

Re: SIP Congestion

Post 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
may the force be with you....
ranjithvkumar

Re: SIP Congestion

Post 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 ?
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cavagnaro
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Post by cavagnaro »

Check logs please. Answer has been given already. There is no magic answer for generic questions
haroun
Senior Member
Posts: 1454
Joined: 29 Mar 2010 11:09

Re: SIP Congestion

Post 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 .
haroun
Senior Member
Posts: 1454
Joined: 29 Mar 2010 11:09

Re: SIP Congestion

Post by haroun »

HI any update with wireshark
do you see g711 rtp or something else?
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